Re: [asterisk-users] bug in Asterisk 1.4.22?
Abel Monzon wrote: and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' Abel, While this is not an a2billing mailing list and you should get more help in their forum, my guess is that the path to the php cli executable is incorrect in the a2billing.php. It's in the first line of it. HTH, Vahan ___ -- 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] bug in Asterisk 1.4.22?
Hi First off, you replied a previous mail to the list, and hence your message appears as part of a previous thread. To post a new message start a new message. Also, On Sun, Oct 26, 2008 at 01:47:03AM -0400, Abel Monzon wrote: Hello is my idea or this is a bug? The thing is that I have in my asterisk.conf this: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /usr/local/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk where the dir of agi-bin is in /usr/local/share/asterisk/agi-bin and inside agi-bin directory I have a file called a2billing.php and in my extesions.conf i have: [a2billing] exten = 1,1,answer exten = 1,2,Wait,2 exten = 1,3,DeadAgi,a2billing.php exten = 1,4,Wait,2 exten = 1,5,Hangup and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory Can you execute it from the shell? ls /usr/local/share/asterisk/agi-bin/a2billing.php /usr/local/share/asterisk/agi-bin/a2billing.php As someone mentioned, if /usr/local/share/asterisk/agi-bin/a2billing.php is there, the error is likely for the executable in the first line (after the #!). -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' So, i change the file a2billing.php to another place and I change this new place in asterisk.conf: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /new/place/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk I reload the asterisk server and the asterisk log still say me the same place before: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory Asterisk reads asterisk.conf only at startup. You'll have to fully restart it. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] bug in Asterisk 1.4.22?
Also check the file permissions and if you are using a RedHat like OS, check the SELinux. And about using a2billing,I recommend you to use version 1.4.21 or less. On Sun, Oct 26, 2008 at 3:30 AM, Tzafrir Cohen [EMAIL PROTECTED]wrote: Hi First off, you replied a previous mail to the list, and hence your message appears as part of a previous thread. To post a new message start a new message. Also, On Sun, Oct 26, 2008 at 01:47:03AM -0400, Abel Monzon wrote: Hello is my idea or this is a bug? The thing is that I have in my asterisk.conf this: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /usr/local/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk where the dir of agi-bin is in /usr/local/share/asterisk/agi-bin and inside agi-bin directory I have a file called a2billing.php and in my extesions.conf i have: [a2billing] exten = 1,1,answer exten = 1,2,Wait,2 exten = 1,3,DeadAgi,a2billing.php exten = 1,4,Wait,2 exten = 1,5,Hangup and then in my softphone I call to 1 the asterisk log say this: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory Can you execute it from the shell? ls /usr/local/share/asterisk/agi-bin/a2billing.php /usr/local/share/asterisk/agi-bin/a2billing.php As someone mentioned, if /usr/local/share/asterisk/agi-bin/a2billing.php is there, the error is likely for the executable in the first line (after the #!). -- Executing [EMAIL PROTECTED]:4] Wait(SIP/abel-28c18000, 2) in new stack == Spawn extension (default, 1, 4) exited non-zero on 'SIP/abel-28c18000' So, i change the file a2billing.php to another place and I change this new place in asterisk.conf: [directories] astetcdir = /usr/local/etc/asterisk astmoddir = /usr/local/lib/asterisk/modules astvarlibdir = /usr/local/share/asterisk astdatadir = /usr/local/share/asterisk astagidir = /new/place/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk I reload the asterisk server and the asterisk log still say me the same place before: -- Launched AGI Script /usr/local/share/asterisk/agi-bin/a2billing.php == a2billing.php: Failed to execute '/usr/local/share/asterisk/agi-bin/a2billing.php': No such file or directory Asterisk reads asterisk.conf only at startup. You'll have to fully restart it. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 -- Juan E. Rodríguez Cel. 829-886-5565 Work: 809-724-9227 ___ -- 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] Cheapest 4 port FXO
On Sat, 25 Oct 2008, Joseph L. Casale wrote: X100P. Yeah I saw these but they are single port and I need at least 2 ports. I only have 1 free pci slot as well. OpenVox. Gordon ___ -- 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] fax / t38 gateway
2008/10/24 Wilton Helm [EMAIL PROTECTED] I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. Fax machines are very easy to use, compared to scanners : you slide a document in, you type a couple of digits and then press send button For most scanners or MFP, you can scan a document sliding it in just like you would when faxing it, but the trouble would be that, frequently, you don't have an easy numeric pad to type your callee's number or address : that's why users don't like to use scanner to send documents. ___ -- 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] Strange ring tone: Long-Short-Short
Joseph wrote: I'm using Linksys SPA3102 adapter and have a strange ring tone: Long-Short-Short or Long-Long-Short-Short Does anybody know which setting adjust this ring tone on SPA3102 Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab Interestingly, I get that, too... but only SOMEtimes. I swear, the number of weird issues I've had with the Linksys ATAs is staggering -- occasionally losing all their stored configs, sometimes refusing to set an IP either via DHCP or manually, weird rings, etc. This has happened on at least a dozen of them, too. It's a wonder I keep buying the things, but unfortunately, they have the reputation as being the 'best' out there. Kind of sad. It's almost certainly going to be somewhere under the regional tab in one of the distinctive ring areas. But since mine are default, and I get weird patterns only sometimes, I'm hesitant to tell you what values are proper there. N. ___ -- 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] Cheapest 4 port FXO
On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? If its for reliability, i wouldn't recommend x100p's Have a look at ata's. Either four sipura/linksys/cisco 3102 or their eight port version. You can put those tiny boxes directly behind your phone/fax. hw ___ -- 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] fax / t38 gateway
Olivier wrote: 2008/10/24 Wilton Helm [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] I've been following this thread and trying to sort out what is wanted, what is available, and why. Comments to the following would be appreciated and might be useful to others. 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. Fax machines are very easy to use, compared to scanners : you slide a document in, you type a couple of digits and then press send button For most scanners or MFP, you can scan a document sliding it in just like you would when faxing it, but the trouble would be that, frequently, you don't have an easy numeric pad to type your callee's number or address : that's why users don't like to use scanner to send documents. Interestingly, a lot of MFPs have FAX facilities, but most only support FAXing through a PSTN connexion. Many MFPs have an ethernet port, so they can talk directly to the internet. Pretty much all others have a USB port so they could talk to the internet through an attached PC. However, only a few support T.37 or T.38 to allow direct image transmission to other FAX machines on the internet. Even the big floor standing office MFPs typically only offer T.37 or T.38 only through an expensive option card. Whether this is a response to market demand, a chicken and egg there's nothing out there to talk to issue, or something manipulative I'm not sure. Regards, Steve ___ -- 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
[asterisk-users] jingle/gtalk still very troubling
Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/[EMAIL PROTECTED] [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't respond, I don't know how to. Remember I use asterisk only, no soft- or hardphone. Does anyone have suggestions, where to look, what to try? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Strange ring tone: Long-Short-Short
I had the same problem and fixed it with this change: Yes, under the Regional Tab, then under Ring and Call Waiting Tone Spec, then under Ring Waveform: I change it from: Trapezoid to Sinusiod. Now the inbound calls to the FXS ring with a more US ring cadence. Hope this helps. eric84 On Sun, Oct 26, 2008 at 6:36 AM, SIP [EMAIL PROTECTED] wrote: Joseph wrote: I'm using Linksys SPA3102 adapter and have a strange ring tone: Long-Short-Short or Long-Long-Short-Short Does anybody know which setting adjust this ring tone on SPA3102 Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab Interestingly, I get that, too... but only SOMEtimes. I swear, the number of weird issues I've had with the Linksys ATAs is staggering -- occasionally losing all their stored configs, sometimes refusing to set an IP either via DHCP or manually, weird rings, etc. This has happened on at least a dozen of them, too. It's a wonder I keep buying the things, but unfortunately, they have the reputation as being the 'best' out there. Kind of sad. It's almost certainly going to be somewhere under the regional tab in one of the distinctive ring areas. But since mine are default, and I get weird patterns only sometimes, I'm hesitant to tell you what values are proper there. N. ___ -- 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] jingle/gtalk still very troubling
Hi Julien, Gtalk channels work with GoogleTalk clients. Empathy (based on the Telepathy framework) has a Gtalk implementation that is reported to work with Asterisk, too. Jingle channels should work with other Jingle implementations, but there are only a few of them around. One reason is that the Jingle specifications are not yet standardized. We try to keep Asterisk's Jingle implementation as close to the spec as possible though. Work is being done by the Telepathy guys on this area too. I've set up a publicly accessible Jingle Asterisk server, reachable at [EMAIL PROTECTED] Subscribe to this JID's presence status and you'll get automatically registered, you can then place Jingle calls to an echo server. Cheers, Philippe On Sun, Oct 26, 2008 at 1:25 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! I just tried to call a friend using jingle, but I got refused. Errorcode was 502, he tried to call me, heard it ringing once and then it stopped. I used: originate jingle/gtalk_account/[EMAIL PROTECTED] [application] I'm registered to googletalk, but this should mean no harm, or should it. Once I was able to receive a text-message from him, but couldn't respond, I don't know how to. Remember I use asterisk only, no soft- or hardphone. Does anyone have suggestions, where to look, what to try? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] jingle/gtalk still very troubling
Hello Philippe! Do I need a googletalk client? Or can I just use asterisk's originate CLI command? I was under the illusion I could. Otherwise it's a bit problematic. I canonly use text-based applications and they better support JACK audio Connection Kit, for my soundcard is not simple standard. I had problems with that before. Do I need to especially configure my firewall, besides opening all outbound ports? I'm in a small local network, so do I also have to configure port-forwarding. As I said: we succeeded in sending me a text-message, but audio won't work. Signalling is fine, but then establishing the connection always failed. Kindest regards and thanks so far Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] jingle/gtalk still very troubling
The originate command should work. Make sure that the user you're placing the Gtalk/Jingle call is in the buddy list and has Jingle capabilities. The 'jabber show buddies' command will give you that info. Cheers! Philippe On Sun, Oct 26, 2008 at 3:57 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hello Philippe! Do I need a googletalk client? Or can I just use asterisk's originate CLI command? I was under the illusion I could. Otherwise it's a bit problematic. I canonly use text-based applications and they better support JACK audio Connection Kit, for my soundcard is not simple standard. I had problems with that before. Do I need to especially configure my firewall, besides opening all outbound ports? I'm in a small local network, so do I also have to configure port-forwarding. As I said: we succeeded in sending me a text-message, but audio won't work. Signalling is fine, but then establishing the connection always failed. Kindest regards and thanks so far Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] jingle/gtalk still very troubling
Hi! There's something strange. I have entered a couple of buddies. On has Jingle capability and two have resources (Home and Telepathy), but my own account does have no resource, I put myself in the buddies list. Is tat supposed to be? And again about those ports: Accept the 5222 port, do all the other necessary ports have to be opened from the outside (or requested from there) or are they opened from my end? And if they need to be opened from the outside: whichports do I have to open in the firewall (taken from the rtp.conf or is there a range simply given by some standard? Kindest regards and thanks Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Cheapest 4 port FXO
OpenVox. Gordon Appreciate that pointer, those are fairly cheap! Thanks, jlc ___ -- 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] fax / t38 gateway
Steve Underwood [EMAIL PROTECTED] writes: That list rather poorly supports your argument. The PAP2 and the PAP2T do *not* support T.38, despite numerous arguments you'll find to the contrary. Personally I believe Linksys, the manual, and the menus. The manuals and the menus for PAP2T talk about T.38. I haven't tested it, because we use Asterisk 1.2 for our PRI gateways. Hopefully I can test 1.6 with a PRI card soon. /Benny ___ -- 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] fax / t38 gateway
Daniel Hazelbaker [EMAIL PROTECTED] writes: I can answer both of those with a single point. We just switched (entirely) to Asterisk a few weeks ago. We looked, very briefly, at various ways to get rid of the physical, analog, fax machines. They all ended with the answer People can't figure out e-mail as it is, they aren't going to figure out how to fax via e-mail.. What we need is a pure VoIP fax machine. HP's and Brothers can do T.37. Unfortunately making them do T.37 means teaching the users a completely different user interface. All that is needed is a way to tell the fax machines that when a user types 123 456 7890 on the fax machines and presses start, it should act as if the machine had switched to the alphabetic keyboard, gone through a bunch of menus, and typed [EMAIL PROTECTED] /Benny ___ -- 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] jingle/gtalk still very troubling
Hi Julien, On Sun, Oct 26, 2008 at 4:51 PM, Julien Claassen [EMAIL PROTECTED] wrote: Hi! There's something strange. I have entered a couple of buddies. On has Jingle capability and two have resources (Home and Telepathy), but my own account does have no resource, I put myself in the buddies list. Is tat supposed to be? The account Asterisk connects with (in jabber.conf) appears in the buddy list, with a default resource named 'asterisk', and has Jingle capabilities. Usually, when you see a buddy without any resource, it means that this buddy is in your roster, but is not currently connected. And again about those ports: Accept the 5222 port, do all the other necessary ports have to be opened from the outside (or requested from there) or are they opened from my end? And if they need to be opened from the outside: whichports do I have to open in the firewall (taken from the rtp.conf or is there a range simply given by some standard? Gtalk and Jingle channels use Asterisk's RTP stack. The UDP port is negociated and can take any value, in the range specified in rtp.conf for Asterisk, unknown for the remote peer. Philippe ___ -- 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] fax / t38 gateway
Steve Underwood [EMAIL PROTECTED] writes: Even the big floor standing office MFPs typically only offer T.37 or T.38 only through an expensive option card. Medium MFP's almost all support T.37. They call it scan to email, but they do it (as far as I can tell) in a way that is compliant with T.37. The user interface is useless though. /Benny ___ -- 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] Cheapest 4 port FXO
In this application what are the pros and cons of using a multiport ata vs a tdm400/800/2400? Eric On Sat, Oct 25, 2008 at 10:54 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? Thanks! jlc ___ -- 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] jingle/gtalk still very troubling
Well, so asterisk seems to think, that I'm not connected, for I don't see a resource Asterisk or Talk with my name. That shouldn't really be. :-( Any ideas on fixing this? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Strange ring tone: Long-Short-Short
What influence the ring tone and patterns is the setting under Regional Tab Ring1 Cadence: 60(2/4) (this is default setting) and selections under User 1 Tab Default Ring: 1 I was thinking about it yesterday Sinusoid vs. Trapezoid setting as all the units I have several old Sipura unit and few Linksys units and they are all set to Trapezoid. I don't remember the reason why I set them this way but I had some problem with Sinusoid setting on Sipura units. All my old-Sipura ring normally with Trapezoid setting. I'll try to change it back to Sinusoid on LinkSys and will confirm later on the outcome (as the unit is not in my location). As Sip in his post has mentioned, I have experienced some strange problem with Linksys units as well. The old-Sipura units maybe they didn't have extra functionality like router etc but they are working without any incident for several years much better then the current LinkSys unit. So the old solid reputation of Sipura should not apply to current LinkSys units. Linksys bought Sipura and bugger them up in addition LinkSys technical support is very poor. -- #Joseph On 10/26/08 10:19, Eric Moniz wrote: I had the same problem and fixed it with this change: Yes, under the Regional Tab, then under Ring and Call Waiting Tone Spec, then under Ring Waveform: I change it from: Trapezoid to Sinusiod. Now the inbound calls to the FXS ring with a more US ring cadence. Hope this helps. eric84 On Sun, Oct 26, 2008 at 6:36 AM, SIP [EMAIL PROTECTED] wrote: Joseph wrote: I'm using Linksys SPA3102 adapter and have a strange ring tone: Long-Short-Short or Long-Long-Short-Short Does anybody know which setting adjust this ring tone on SPA3102 Sipura rings normally. I'm not sure if setting are on Regional Tab or User Tab Interestingly, I get that, too... but only SOMEtimes. I swear, the number of weird issues I've had with the Linksys ATAs is staggering -- occasionally losing all their stored configs, sometimes refusing to set an IP either via DHCP or manually, weird rings, etc. This has happened on at least a dozen of them, too. It's a wonder I keep buying the things, but unfortunately, they have the reputation as being the 'best' out there. Kind of sad. It's almost certainly going to be somewhere under the regional tab in one of the distinctive ring areas. But since mine are default, and I get weird patterns only sometimes, I'm hesitant to tell you what values are proper there. N. -- #Joseph GPG KeyID: ED0E1FB7 ___ -- 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] jingle/gtalk still very troubling
Strange, are you both connected to Google's XMPP server? Sometimes it takes a little time before retrieving your roster on Gtalk. Does Asterisk appear as connected on your friend's buddy list? Also, what does the 'jabber show connected' say? Cheers, Philippe On Sun, Oct 26, 2008 at 5:53 PM, Julien Claassen [EMAIL PROTECTED] wrote: Well, so asterisk seems to think, that I'm not connected, for I don't see a resource Asterisk or Talk with my name. That shouldn't really be. :-( Any ideas on fixing this? Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- Philippe Sultan ___ -- 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] Cheapest 4 port FXO
in multiport sipura/Linksys you cannot access individual ports you have to address them by the group _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Fort Sent: Sunday, October 26, 2008 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cheapest 4 port FXO In this application what are the pros and cons of using a multiport ata vs a tdm400/800/2400? Eric On Sat, Oct 25, 2008 at 10:54 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? Thanks! jlc ___ -- 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
[asterisk-users] No incoming audio on Dahdi channels (TDM410P)
A previous issue has popped up and once again I'm out of ideas. During the evenings it seems that the TDM channels will spike (dahdi_monitor) and will refuse to listen for audio of any type, this includes DTMF. The only resolution I know of is to stop Asterisk and restart the dahdi service, but that's not a solution. All channels look like this, even the FXS. [EMAIL PROTECTED] Hardware]# dahdi_monitor 1 -vv Visual Audio Levels. Use chan_dahdi.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX (TX ###* Rx: 30076 (30076) Tx: 0 (0) I've stopped every service except SSH and networking (according to service --status-all) and nothing has changed. [EMAIL PROTECTED] cat /proc/interrupts CPU0 0: 77924086IO-APIC-edge timer 1: 3IO-APIC-edge i8042 6: 6IO-APIC-edge floppy 7: 0IO-APIC-edge parport0 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 12: 4IO-APIC-edge i8042 14: 104093IO-APIC-edge ide0 15: 690398IO-APIC-edge ide1 201: 77835719 IO-APIC-level wctdm24xxp0 209: 770795 IO-APIC-level eth1 NMI: 0 LOC: 77927794 ERR: 0 MIS: 0 Nothing looks shared, but then I see this in lspci -vb: 00:02.0 VGA compatible controller: Intel Corporation 82845G/GL[Brookdale-G]/GE Chipset Integrated Graphics Device (rev 03) (prog-if 00 [VGA controller]) Subsystem: Micro-Star International Co., Ltd. Unknown device 5578 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at d000 (32-bit, prefetchable) Memory at dff8 (32-bit, non-prefetchable) Capabilities: [d0] Power Management version 1 ... ... 01:01.0 Ethernet controller: Digium, Inc. Unknown device 8005 (rev 11) Subsystem: Digium, Inc. Unknown device 8005 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at cc00 Memory at dfdffc00 (32-bit, non-prefetchable) Expansion ROM at dfdc [disabled] Capabilities: [c0] Power Management version 2 Is that normal? Here's the output of dahdi_diag 1: dahdi: Dump of DAHDI Channel 1 (WCTDM/0/0,1,1): dahdi: flags: 201 hex, writechunk: ee0d008c, readchunk: ee0d0098 dahdi: rxgain: f8b8c480, txgain: f8b8c480, gainalloc: 0 dahdi: span: e9460054, sig: 2004 hex, sigcap: 6085 hex dahdi: inreadbuf: -1, outreadbuf: -1, inwritebuf: -1, outwritebuf: -1 dahdi: blocksize: 0, numbufs: 2, txbufpolicy: 0, txbufpolicy: 0 dahdi: txdisable: 0, rxdisable: 0, iomask: 0 dahdi: curzone: , tonezone: 0, curtone: , tonep: 0 dahdi: digitmode: 0, txdialbuf: , dialing: 0, aftdialtimer: 0, cadpos. 0 dahdi: confna: 0, confn: 0, confmode: 0, confmute: 0 dahdi: ec: , echocancel: 0, deflaw: 0, xlaw: f8b6f2a0 dahdi: echostate: 00, echotimer: 0, echolastupdate: 0 dahdi: itimer: 0, otimer: 0, ringdebtimer: 0 No idea what any of that means or how it's relevant. dmesg is full of interrupt misses and polarity reversals: ... wctdm24xxp0: Missed interrupt. Increasing latency to 18 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 19 ms in order to compensate. 29794979 Polarity reversed (1 - -1) 29795839 Polarity reversed (-1 - 1) wctdm24xxp0: Missed interrupt. Increasing latency to 20 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 21 ms in order to compensate. wctdm24xxp0: Missed interrupt. Increasing latency to 22 ms in order to compensate. 31595924 Polarity reversed (1 - -1) 31596867 Polarity reversed (-1 - 1) ... RING on 1/2! 74920374 Polarity reversed (-1 - 1) NO RING on 1/2! 74921961 Polarity reversed (1 - -1) RING on 1/2! NO RING on 1/2! NO BATTERY on 1/2! BATTERY on 1/2 (-)! Running AsteriskNow 1.5. X Windows is disabled. Ideas? Suggestions? Thoughts? Going to build another PC and toss this in there to see what happens tonight. Thanks. ___ -- 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] jingle/gtalk still very troubling
Evening Philippe! Here's what jabber show connected says: Jabber Users and their status: User: [EMAIL PROTECTED]/Talk - Connected Number of users: 1 I'll have to ask my friends, what their clients say. Although I suppose as my friend already send me a text message he saw me. And the state of me having no resource still appears after hours of running asterisk, with no change in configuration. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Cheapest 4 port FXO
Sunday, October 26, 2008, 12:31:16 PM, Hans wrote: On Sat, 2008-10-25 at 11:54 -0600, Joseph L. Casale wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? If its for reliability, i wouldn't recommend x100p's Have a look at ata's. Either four sipura/linksys/cisco 3102 or their eight port version. You can put those tiny boxes directly behind your phone/fax. Well, there're linksys SPA400 what have 4 FXO. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ -- 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
[asterisk-users] Asterisk on Freebsd 7.0 Release.
Hello, I want to know if some body use Asterisk on Freebsd 7.0 release? My problem is that, when I call to any extension and the asterisk need to reproduced a file GSM o MP3, whatever, that have a lot of noise... Only not have noise when the extensions is not avalaible. That happen only in Freebsd 7.0, on Windows 32 don't happend that, Debian either... so, some body know why is that? Thanks, Abel Monzon___ -- 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] Fresh installed box
Thanks Matt, I will check them. From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sat, 25 Oct 2008 14:28:27 -0400 Subject: Re: [asterisk-users] Fresh installed box Hi Torintino, 1. Login to FreePBX, Go to extensions, Select the extension you want to configure, Scroll down to the bottom under the voicemail setup section, and check the “Attach to Email” checkbox and then save the extension and reload freepbx. Now your emails will be sent including the voicemail. Note that mail has to be setup on the box for it to work (ssmtp or local mta). 2. Here are some tutorials - http://www.voip-info.org/wiki-Asterisk+fax, http://www.voip-info.org/wiki/view/T.38 http://nerdvittles.com/index.php?p=88 http://asterfax.sourceforge.net/ 3. Ah, I’m not positive on what would work for this – sounds like some modifications to FOP may be in need. Maybe someone else on the list has ideas. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Torintino T Sent: Saturday, October 25, 2008 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box Thanks Matt, would you please tell me in details about the following 1- the Linux mail configuration steps to enable it to send voicemail to email. 2- the steps to use T.38 and pass thru...or Fax detection...and fax to email. 3- for the live monitoring.i wanna a software to monitor and to make spying on the calls, etc... if you will send me helpful documents , your help will be appreciated Thanks, Torintino From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Oct 2008 21:45:22 -0400 Subject: Re: [asterisk-users] Fresh installed box http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 24, 2008 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: Matt Gibson [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- 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 Discover the new Windows Vista Learn more! _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- 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] Cheapest 4 port FXO
On Sun, 26 Oct 2008, Eric Fort wrote: In this application what are the pros and cons of using a multiport ata vs a tdm400/800/2400? For me it would be much easier to access the on-board TDM card and less wiring/mains units. (power units) Gordon Eric On Sat, Oct 25, 2008 at 10:54 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I need to increase reliability at an office as SIP/Internet provider outages are causing some issues. What would be the least expensive analogue card that people are using reliably? Thanks! jlc ___ -- 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] Cheapest 4 port FXO
On Sun, Oct 26, 2008 at 4:51 AM, Gordon Henderson [EMAIL PROTECTED] wrote: On Sat, 25 Oct 2008, Joseph L. Casale wrote: X100P. Yeah I saw these but they are single port and I need at least 2 ports. I only have 1 free pci slot as well. OpenVox. Those look great, and on top of the price they are 100% TDM400P compatible it seems. ___ -- 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] OT: Disable Polycom 650 Forward Softkey
On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote: The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? Which would be great, if Polycom weren't the Firmware-Nazis that they are. ___ -- 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] Emerging dilema? DID forwarding meets SMS
On Fri, Oct 24, 2008 at 10:09 AM, Drew Gibson [EMAIL PROTECTED] wrote: Can anyone clarify how SMS to non-mobile numbers are generally handled in North America? Is it possible to have SMS delivered direct to your landline DIDs? Then have Asterisk relay it to the actual mobile DID. When I send an SMS from a SprintPCS phone to a landline it gets delivered via voice, pretty much how Gordon describes ___ -- 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] OT: Disable Polycom 650 Forward Softkey
If you buy your phone from a reputable place they will be able to provide the firmware. --Original Message-- From: Andrew Joakimsen Sender: To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey Sent: Oct 26, 2008 5:45 PM On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote: The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? Which would be great, if Polycom weren't the Firmware-Nazis that they are. ___ -- 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 Sent from my BlackBerry® wireless device from U.S. Cellular ___ -- 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] fax / t38 gateway
Benny Amorsen wrote: Steve Underwood [EMAIL PROTECTED] writes: That list rather poorly supports your argument. The PAP2 and the PAP2T do *not* support T.38, despite numerous arguments you'll find to the contrary. Personally I believe Linksys, the manual, and the menus. The manuals and the menus for PAP2T talk about T.38. I haven't tested it, because we use Asterisk 1.2 for our PRI gateways. Hopefully I can test 1.6 with a PRI card soon. A number of people say this, but when you ask them to point out the exact location in the menus, they can't find it. There does appear to have been a version 3.something firmware with half finished a buggy T.38 implementation. None of the earlier or later versions include it. I have not seen a manual which mentions it. There are release notes which mention it, but when you look carefully they are generic notes and the T.38 part does not apply to the PAP2T. The amount of folklore surrounding this is kinda frustrating when you are looking for people to help test T.38. :-) Regards, Steve ___ -- 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] fax / t38 gateway
Benny Amorsen wrote: Steve Underwood [EMAIL PROTECTED] writes: Even the big floor standing office MFPs typically only offer T.37 or T.38 only through an expensive option card. Medium MFP's almost all support T.37. They call it scan to email, but they do it (as far as I can tell) in a way that is compliant with T.37. The user interface is useless though. Not most, just some, and they don't utually do it out of the box. The Brothers, for example, require a clumsy download and update that most people will never be able to cope with. Most of those implementations are also quirky, and do not comply with T.37. They also dumb down the documentation to the point where it can be unusable. Regards, Steve ___ -- 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] Panasonic x Asterisk ... NO PROBLEM!
You could, under programming section 1.3.4 in the http interface to configure the GW card enable DTMF Detection, that will enable Out of Band DTMF. In the TDE they renamed this to DTMF signalling. On Fri, Oct 24, 2008 at 2:42 PM, Richard Scobie [EMAIL PROTECTED] wrote: Jonn R Taylor wrote: Install a T1 between the Panasonic and Asterisk and program the T1 in the Panasonic as a other custom PBX. VOIP card would be the best. Jonn One thing to beware of with the Panasonic VoIP card, is that I have found no way of getting it to pass out of band DTMF, possibly because it handles this in a proprietary way. This has been my experience with a TDA100 and VoIP card. Regards, Richard ___ -- 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
[asterisk-users] CDR Records are not working
Hello Asterisk-Users,
For some reason my CDR records for disposition and billsec are not working
correctly.
I always receive a 0 for billsec and the disposition is always at NO
ANSWER', even when I grab the calls.
I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22.
Here is information on how I do the call:
-
.call file contents:
-
Channel: SIP/GAFACHI/1818345
CallerID: 18183455512
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: outboundmessage1
Extension: s
Priority: 1
Set: PassedInfo=18183453041-m1d
-
extensions.conf for outboundmessage1 context:
-
[outboundmessage1]
exten = s,1,Set(CDR(userfield)=${PassedInfo})
exten = s,2,Answer
exten = s,3,System(/opt/asterisk/scripts/custom/answer.sh ${CDR(clid)}
${CDR(dst)} ${CDR(billsec)})
exten = s,4,Background(/tmp/hello-world)
exten = s,5,WaitExten()
-
Master.csv after picking up the line:
-
,1818345512,2,outboundmessage1,1818345,SIP/GAFACHI-090cd790,,Hangup,,2008-10-27
00:19:08,,2008-10-27 00:19:33,25,0,NO
ANSWER,DOCUMENTATION,1225066748.1,18183453041-m1d
Any insight would be appreciated.
Thanks,
Pedram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Records are not working
I have the same problem for Disposition when I use call files. The
duration is correct but the Disposition is always NO ANSWER. I also am
using 1.6.0.1. I did not have the problem when I was using 1.4.21
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pedram M
Sent: Monday, 27 October 2008 10:46 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] CDR Records are not working
Hello Asterisk-Users,
For some reason my CDR records for disposition and billsec are not
working correctly.
I always receive a 0 for billsec and the disposition is always at NO
ANSWER', even when I grab the calls.
I experience this with Asterisk 1.6.0.1 and Asterisk 1.4.22.
Here is information on how I do the call:
-
.call file contents:
-
Channel: SIP/GAFACHI/1818345
CallerID: 18183455512
MaxRetries: 0
RetryTime: 60
WaitTime: 30
Context: outboundmessage1
Extension: s
Priority: 1
Set: PassedInfo=18183453041-m1d
-
extensions.conf for outboundmessage1 context:
-
[outboundmessage1]
exten = s,1,Set(CDR(userfield)=${PassedInfo})
exten = s,2,Answer
exten = s,3,System(/opt/asterisk/scripts/custom/answer.sh ${CDR(clid)}
${CDR(dst)} ${CDR(billsec)})
exten = s,4,Background(/tmp/hello-world)
exten = s,5,WaitExten()
-
Master.csv after picking up the line:
-
,1818345512,2,outboundmessage1,1818345,SIP/GAFACHI-090cd7
90,,Hangup,,2008-10-27 00:19:08,,2008-10-27 00:19:33,25,0,NO
ANSWER,DOCUMENTATION,1225066748.1,18183453041-m1d
Any insight would be appreciated.
Thanks,
Pedram
___
-- 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
[asterisk-users] Asterisk and voice recognition
Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- 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] OT: Disable Polycom 650 Forward Softkey
Other vendors, including Cisco, will provide the firmware directly. I no longer deploy Polycom (unless someone really wants them) due to this. Yes I can get it from the supplier but it takes a few days. I would rather just go to Polycom.com and get the firmware when I want to. There is no excuse for Polycom's behaviour. I don't see what is the benefit, nor what anyone has to gain from it. On Sun, Oct 26, 2008 at 7:53 PM, Darrick Hartman [EMAIL PROTECTED] wrote: If you buy your phone from a reputable place they will be able to provide the firmware. --Original Message-- From: Andrew Joakimsen Sender: To: Asterisk Users Mailing List - Non-Commercial Discussion ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT: Disable Polycom 650 Forward Softkey Sent: Oct 26, 2008 5:45 PM On Fri, Oct 24, 2008 at 10:19 PM, Chris Walton [EMAIL PROTECTED] wrote: The 3.1.0 firmware allows you to create up to 10 custom softkeys. This is all documented in Polycom's SIP 3.1 Admin Guide. Should I post some examples? Which would be great, if Polycom weren't the Firmware-Nazis that they are. ___ -- 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 and voice recognition
Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- 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 and voice recognition
Hi, Many thanks for that info. Is there any free solution available as well? Many thanks, Christian On 2008-10-26 at 20:32 Darren Sessions wrote: Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- 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 ___ -- 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
[asterisk-users] Asterisk 1.6 CDR no Clid information
Hi All, For some reason since moving to Asterisk 1.6. my CDR records are no longer displaying the Clid field. The CDR records contain the Source field be for some reason not the CID details. I am logging CDR to mysql. Is anyone able to help? Regards David. ___ -- 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 and voice recognition
Sphinx http://cmusphinx.sourceforge.net/html/cmusphinx.php Not sure how the implementation works with Asterisk but I know it's been done (I'd google it). - D On 26 Oct 2008, at 20:55, Christian wrote: Hi, Many thanks for that info. Is there any free solution available as well? Many thanks, Christian On 2008-10-26 at 20:32 Darren Sessions wrote: Not sure about the Swedish, but Lumenvox has a great speech recognition app for Asterisk. - D On 26 Oct 2008, at 19:53, Christian wrote: Hi all, Yes, this might not be the proper list for this, but i have a question about Asterisk and voice recognition. If I want to create a menu system where the user can say different things in the Swedish language what should I look at? For example, i want the user to be able to say something simular in Swedish: connect disconnect help and so on. Best regards and thanks, Christian ___ -- 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 ___ -- 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] OT: Disable Polycom 650 Forward Softkey
On Sunday 26 October 2008 21:28:34 Andrew Joakimsen wrote: Other vendors, including Cisco, will provide the firmware directly. I no longer deploy Polycom (unless someone really wants them) due to this. Yes I can get it from the supplier but it takes a few days. I would rather just go to Polycom.com and get the firmware when I want to. There is no excuse for Polycom's behaviour. I don't see what is the benefit, nor what anyone has to gain from it. I believe your anger is misplaced. I was able to get to a direct download of Polycom firmware, from their homepage, within 4 clicks, with no login whatsoever required. http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip501.html While Polycom at one time may have had a policy of only providing firmware to distributors and resellers, that is no longer the case. Their firmware is freely available now to all comers. -- Tilghman ___ -- 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] Panasonic x Asterisk ... NO PROBLEM!
C F wrote: You could, under programming section 1.3.4 in the http interface to configure the GW card enable DTMF Detection, that will enable Out of Band DTMF. In the TDE they renamed this to DTMF signalling. Believe me, I spent a great deal of time on this including Ethereal captures and nothing worked. Have you succeeded with this? If so, what DTMF protocols were passed? Regards, Richard ___ -- 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
[asterisk-users] autodialed call forwarding via meetme or queue (was predictive dialer)
Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the agents notice that the other line is silent. The queue is already set up to hold an infinite number of calls (meaning: maxlen=0/no limit), and the agents are already answering the calls immediately/after one ring, but the problem still shows up. Is forwarding to a meetme conference room faster than through a queue? On Thu, Oct 16, 2008 at 11:25 PM, Steve Totaro [EMAIL PROTECTED] wrote: If you can figure out how to generate .call files from your DB entries, you have it made. Vicidial needs alot of work as far as I am concerned, for free it is OK I guess. I think using meetme conference rooms for everything is a kludgy hack, and the UI is less than nice (if you are into UIs). I suggest you continue on your own custom development if you have the time. Check out Aheeva for inspiration. Thanks, Steve Totaro On Fri, Oct 17, 2008 at 1:31 AM, ram [EMAIL PROTECTED] wrote: look at Vicidial ram On Thu, Oct 16, 2008 at 4:46 PM, yavuz yildirim [EMAIL PROTECTED] wrote: hi everybody This is Yavuz YILDIRIM I am software developer.I have a some problems in asterisk. I am using mysql db. Realtime using asterisk modules. On db i am using calling hundred fields for use dial. But i don't know how i can automaticly dial this fields on records numbers. Who can help me asterisk api and others. Thank you ___ -- 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 -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
