[Asterisk-Users] Asterisk TAPI - Outlook click2dial
Hello, I'm experiencing some problems with AstTAPI driver. Dialing works just fine, but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact that Outlook doesen't detect end of conversation -> once the call is terminated 'manually' via the phone Outlook still 'thinks' that call is active. Anyone knows what's the problem? Is 'hangup' implemented in AstTAPI driver? Thanks, Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted -> omniis.com. Version is 0.10 (newest) Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup' isn't even implemented in AstTAPI driver so that could be the reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T.S Sent: Thursday, May 11, 2006 1:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial I had similar problems when I first started to play with it. I've gotten Omniis TSP for Astrisk to work just fine. http://www.omniis.com/asttapi But i don't know the version im using 0.0.8 Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Wednesday, May 10, 2006 2:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hello, I'm experiencing some problems with AstTAPI driver. Dialing works just fine, but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact that Outlook doesen't detect end of conversation -> once the call is terminated 'manually' via the phone Outlook still 'thinks' that call is active. Anyone knows what's the problem? Is 'hangup' implemented in AstTAPI driver? Thanks, Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1530 (20060510) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial
Oh.. :/ too bad.. I'll have to look at the source.. bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw Sent: Thursday, May 11, 2006 11:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Yes, I have the exact same problem. :( -Original Message- From: Tomislav Vojvodic [mailto:[EMAIL PROTECTED] Sent: Thursday, May 11, 2006 5:48 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted -> omniis.com. Version is 0.10 (newest) Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup' isn't even implemented in AstTAPI driver so that could be the reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T.S Sent: Thursday, May 11, 2006 1:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial I had similar problems when I first started to play with it. I've gotten Omniis TSP for Astrisk to work just fine. http://www.omniis.com/asttapi But i don't know the version im using 0.0.8 Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Wednesday, May 10, 2006 2:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hello, I'm experiencing some problems with AstTAPI driver. Dialing works just fine, but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact that Outlook doesen't detect end of conversation -> once the call is terminated 'manually' via the phone Outlook still 'thinks' that call is active. Anyone knows what's the problem? Is 'hangup' implemented in AstTAPI driver? Thanks, Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1530 (20060510) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1532 (20060511) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] time update (7905)
Hi everyone, I'm trying to update time on all Cisco 7905 phones in my company.. is there some way to do it from asterisk? Thanks, Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial out .call files File permissions??
I copy/pasted Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 ..and saved it as 'callme' .. ..and put chmod 777 callme ..and mv callme /var/spool/asterisk/outgoing/ All as root - and.. it's working ;) (Tested on AAH 2.7) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 28, 2006 11:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dial out .call files File permissions?? Hi all, I've created this test.call file and it is not running outgoing call files: i've made mv test.call /var/spool/asterisk/outgoing and nothing happens Channel: SIP/200 MaxRetries: 3 RetryTime: 40 WaitTime: 25 Context: from-internal Extension: 200 Priority: 1 My asterisk is running with asterisk user. not root user. Could you help me on ? Could this be a problem of file permissions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial out .call files File permissions??
If you put 777 on some file, that means that anyone can read/write/execute that file.. I think that file ownership isn't important in that case, but if you want to set 'precise' permissions so that only user 'asterisk' can deal with it.. then type chmod 644 filename (owner can read/write, others can only read) ..or chmod 600 filename (only owner can read/write) ..and set file ownership to asterisk user (make sure that asterisk is really running under asterisk user) chown asterisk.asterisk filename --- Just for the record, I'm really new with * so it is possible that what I'm saying isn't correct. This is actually a linux question.. Anywayz, hope this helps. Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruno De Luca Sent: Tuesday, March 28, 2006 12:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial out .call files File permissions?? do u need to give the permission for user asterisk to uour file. Bruno. Marco Mouta wrote: > Hi all, > > I've created this test.call file and it is not running outgoing call files: > > i've made mv test.call /var/spool/asterisk/outgoing and nothing happens > > Channel: SIP/200 > MaxRetries: 3 > RetryTime: 40 > WaitTime: 25 > Context: from-internal > Extension: 200 > Priority: 1 > > My asterisk is running with asterisk user. not root user. > > Could you help me on ? Could this be a problem of file permissions? > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1459 (20060327) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial out .call files File permissions??
heh.. I just noticed that ;) Heh, do you know maybe how to update time/date on all Cisco 7905 phones through asterisk? I need to increase time for 1 hour.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta Sent: Tuesday, March 28, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dial out .call files File permissions?? it's working , the problem was: Channel: ZAP/g1X I changed to ZAP/g1/X And it's working fine! Thank you all On 3/28/06, Marco Mouta <[EMAIL PROTECTED]> wrote: > Thank you for your fast reply!!! > It's working on for SIP:) > > I've tried to my zapata and doesn't make the call, i get this: > > Attempting call on ZAP/g1964391121 for [EMAIL PROTECTED]:1 (Retry 2) > > -- Attempting call on ZAP/[EMAIL PROTECTED] for > [EMAIL PROTECTED]:1 (Retry 1) > -- Attempting call on ZAP/[EMAIL PROTECTED] for > [EMAIL PROTECTED]:1 (Retry 2) > -- Attempting call on ZAP/[EMAIL PROTECTED] for > [EMAIL PROTECTED]:1 (Retry 1) > > Do I need to define context to outbound calls through my ZAP ? > > Thanks in advance, > Marco Mouta > > On 3/28/06, Tomislav Vojvodic <[EMAIL PROTECTED]> wrote: > > I copy/pasted > > > > Channel: SIP/200 > > MaxRetries: 3 > > RetryTime: 40 > > WaitTime: 25 > > Context: from-internal > > Extension: 200 > > Priority: 1 > > > > ..and saved it as 'callme' .. > > > > ..and put chmod 777 callme > > > > ..and mv callme /var/spool/asterisk/outgoing/ > > > > All as root - and.. it's working ;) > > > > (Tested on AAH 2.7) > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Marco Mouta > > Sent: Tuesday, March 28, 2006 11:47 AM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] Dial out .call files File permissions?? > > > > Hi all, > > > > I've created this test.call file and it is not running outgoing call files: > > > > i've made mv test.call /var/spool/asterisk/outgoing and nothing happens > > > > Channel: SIP/200 > > MaxRetries: 3 > > RetryTime: 40 > > WaitTime: 25 > > Context: from-internal > > Extension: 200 > > Priority: 1 > > > > My asterisk is running with asterisk user. not root user. > > > > Could you help me on ? Could this be a problem of file permissions? > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] time update (7905)
Thanks 2 everyone ;) As you said, I found config template for 7905's, and set DST offset in config template mac-addr-here.cnf #Winter Time UTC+1 #TimeZone:1 #Summer Time UTC+2 TimeZone:2 ..and after reboot (**#**) all 7905's are displaying time correctly. Thnx and bye ;) Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Farmer Sent: Tuesday, March 28, 2006 1:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] time update (7905) --- Michiel van Baak <[EMAIL PROTECTED]> wrote: > On 09:16, Tue 28 Mar 06, Tomislav Vojvodic wrote: > > Hi everyone, > > > > I'm trying to update time on all Cisco 7905 phones > in my company.. is there > > some way to do it from asterisk? Don't have any 7905's but on our 7940's you set the DST settings in the SIPDefault.cnf. Then if you have told the phones to use NTP they update automatically. Regards Jon Jon Farmer Telford, Shropshire, UK ___ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre. http://uk.security.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set caller ID for outgoing PRI calls
It seems it's 'normal' behaviour since I heard exactly the same thing happening in Croatia. If caller id is set to some number, telco overrides it to first caller id.. (even if that number belongs to your block (right?)) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: Tuesday, March 28, 2006 2:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Set caller ID for outgoing PRI calls Hallo! Finally we have E1 PRI connected to our Asterisk box. Now I have one question. My internal extensions (_XXX) are SIP phones connected to Asterisk. Our telco routes some public numbers (_71602XX and others) to our Asterisk via E1. Some internal extensions can be reached from outside using public numbers (e.g. 7160234 -> 200), and some others cannot. Everyone can call outside numbers from our network. How can I set caller id to something meaningful when they are calling outside? For example, 201 -> 7160201, 202 -> 3678685, etc. If I send incorrect caller id, my telco overrides it to first number from my block (7160200). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set caller ID for outgoing PRI calls
umm i don't understand anymore ..you have one block of numbers - E1 / PRI and you want to assign number that is not in your block? Why do you want to do that? You have 'external numbers' which you can route to your local extensions and you can set caller id to number from e1-block when someone is calling outside, right? Now what? :) If you have some additional interfaces, for example, 2 BRI with n numbers -you can't mix numbers from different interfaces. You could ask your telco about that.. but.. hm... Is that what you were asking? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: Tuesday, March 28, 2006 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Set caller ID for outgoing PRI calls On Tuesday 28 March 2006 15:40, Tomislav Vojvodic wrote: > It seems it's 'normal' behaviour since I heard exactly the same thing > happening in Croatia. If caller id is set to some number, telco > overrides it to first caller id.. (even if that number belongs to > your block (right?)) No. It sets Caller id to the first number only if it does not belong to my block. This is why I wish to set caller id myself when originating calls from office extensions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Set caller ID for outgoing PRI calls
oh i see ;) i talked to my colleague about this, and he said this should probably need telco intervention to do that for you.. I'm not sure that call forwarding will do the trick correctly with the 'caller id'.. but I think it's all about your telco again ;) Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Tuesday, March 28, 2006 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Set caller ID for outgoing PRI calls Because, sometimes you have an office suite style configuration where you want to push out an 800 # because they get cheaper rates going out the PRI than their other circuit or because you are in the process of porting a # and don't want to confuse people who pick up the phone (and you still want your clients normal phone # to show up) or simply for redundancy reasons - pushing out a call over the 2nd PRI if the first one is full (which may be at a different carrier for redundancy). Another big issue is call forwarding - call comes in the PRI from my cell 301-748- and enters my Asterisk box where it gets forwarded out to say my home phone. Well, Asterisk sets the caller ID to 301-748- but the telco drops it so it shows up at my home phone as the first number of the DID block. How do I know who is calling then? I don't! We just enabled "call redirection" feature that our upstream offers but it's not activated yet - but according to them this will fix that problem. As simple as the concept is to most people, I found that explaining the call forwarding issue usually gets the carrier to understand what we want accomplished. Then again, I don't know if that will work yet. :) Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Tuesday, March 28, 2006 8:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Set caller ID for outgoing PRI calls umm i don't understand anymore ..you have one block of numbers - E1 / PRI and you want to assign number that is not in your block? Why do you want to do that? You have 'external numbers' which you can route to your local extensions and you can set caller id to number from e1-block when someone is calling outside, right? Now what? :) If you have some additional interfaces, for example, 2 BRI with n numbers -you can't mix numbers from different interfaces. You could ask your telco about that.. but.. hm... Is that what you were asking? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: Tuesday, March 28, 2006 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Set caller ID for outgoing PRI calls On Tuesday 28 March 2006 15:40, Tomislav Vojvodic wrote: > It seems it's 'normal' behaviour since I heard exactly the same thing > happening in Croatia. If caller id is set to some number, telco > overrides it to first caller id.. (even if that number belongs to > your block (right?)) No. It sets Caller id to the first number only if it does not belong to my block. This is why I wish to set caller id myself when originating calls from office extensions. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1459 (20060327) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Budgetone and Mac mini?
I don't have any experience with that specific phone, but i have a little experience with Grandstream ATAs. Check web configuration to see is phone in gateway or bridge mode. It's just a guess since you haven't provide detailed info ;) What do you mean when you say that none of them works with mac mini? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dmitry Ivanov Sent: Tuesday, April 18, 2006 11:35 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream Budgetone and Mac mini? Hallo! Anyone tried connect PC port of BT-102 to Mac mini? I have four BT-102. Looks like none of them works with Mac mini G4... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1493 (20060417) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using kannel with asterisk
Well kannel by itself doesen't use much resources as far as I remember.. it's all about actions taken upon receiving sms.. Please let me know your experiences since I'm also interested in kannel / asterisk combination.. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: Thursday, June 29, 2006 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using kannel with asterisk hello I have an asterisk server with a te110p E1 digium card. the server is a hp ML370 3,2 Ghz 64bits, 1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5. I want to use in the same machine the kannel SMSC. i have no big trafic in the two gateway but I want to know if it generate a performence problem for asterisk I use fedora core4 with latest asterisk version . thanks Regards issam __ NOD32 1.1632 (20060629) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using kannel with asterisk
If you'll use newer distribution of linux you'll probably jump into problems with libsqlite3 (libsqlite2 is needed for kannel).. it is well documented on kannel website.. you can contact me off-list about kannel since this isnt't kannel mailing list... I got kannel and asterisk running under CentOS 4.3 with forced sqlite2 install ;) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: Friday, June 30, 2006 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] using kannel with asterisk I don't use asterisk in combination with kannel. Actually we use nowsms as SMSC gateway to connect to our provider but we deside to replace it by kannel. so we store incoming messages in an sqlserver 2005 database in windows 2003 server . please let me what you need to combine kannel and asterisk ? thanks Regards issam - Original Message - From: Tomislav Vojvodic To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, June 29, 2006 11:00 AM Subject: RE: [Asterisk-Users] using kannel with asterisk Well kannel by itself doesen't use much resources as far as I remember.. it's all about actions taken upon receiving sms.. Please let me know your experiences since I'm also interested in kannel / asterisk combination.. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of issam Sent: Thursday, June 29, 2006 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using kannel with asterisk hello I have an asterisk server with a te110p E1 digium card. the server is a hp ML370 3,2 Ghz 64bits, 1Mo L2 , 1Go Ram, 3 SCSI 73Go in raid5. I want to use in the same machine the kannel SMSC. i have no big trafic in the two gateway but I want to know if it generate a performence problem for asterisk I use fedora core4 with latest asterisk version . thanks Regards issam __ NOD32 1.1632 (20060629) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
