Re: [asterisk-users] Asterisk 1.8 Service: -r does not give CLI
Not it didn't, Did you execute asterisk -r or /usr/sbin/asterisk -r ? If not working did you execute asterisk -gc ? Kind Regards On Mon, Jul 22, 2013 at 10:41 AM, Meadows Hoa meadows_...@yahoo.com wrote: We have Asterisk1.8.11 and can not move to a newer version right now. But when we run Asterisk as a service, the -r option does not result in giving the CLI prompt? Did the option to get the CLI change? -- _ -- 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] Which is the stable version to use?
Depends on used kernel and perhaps on other hardware you are using. Am 23.07.2013 00:09, schrieb bilal ghayyad: Hello I need to deploy asterisk on production and same thing for DAHDI, which version is recommended for this? Regards Bilal -- _ -- 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] Set ringtone by dialed number
The reason for this is we have one primary company office but there are two entities and if someone call the Denver number it would be for 1 organization and would ring differently helping our staff remember how to answer the phone for the Denver organization rather than for the Colorado Springs number which is a different entity. A lot of times people are rushing to answer the phone and do not look at the callerID this would give them and auditory reminder of how they need to answer the phone. How would I go about setting up telling the phone to change the ring tone in the SIP header? On Monday 22 July 2013, Josh Hopkins wrote: Would it be possible to set the ringtone based on the number that was dialed? If the phones you are using allow the ringing tone to be changed by sending a SIP header, yes. Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 But why would anyone want this? What is the point of changing the sound that your phone makes when someone calls you, depending on who you called last? We are currently using Digium D40, D50, D70 phones. -- AJS Answers come *after* questions. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Hopkins Sent: Monday, July 22, 2013 9:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Set ringtone by dialed number Would it be possible to set the ringtone based on the number that was dialed? Example of what the goal is: Dial Denver number Incoming calls ring with ringtone 1 Dial main number Incoming calls ring with ringtone 2 We are currently using Digium D40, D50, D70 phones. -- _ -- 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] Set ringtone by dialed number
Josh! This is a typical application for using different SIP identities. Essentially, no programming is involved---just configuration. Hopefully, your SIP phones allow for several identities. In Asterisk you would dispatch the calls and connect to the proper SIP accounts. You would also configure your phones to use a different ring tone for each account. In addition you would also have some visual feedback (line buttons, display text). jg -- _ -- 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] Set ringtone by dialed number
On Tuesday 23 July 2013, Josh Hopkins wrote: The reason for this is we have one primary company office but there are two entities and if someone call the Denver number it would be for 1 organization and would ring differently helping our staff remember how to answer the phone for the Denver organization rather than for the Colorado Springs number which is a different entity. A lot of times people are rushing to answer the phone and do not look at the callerID this would give them and auditory reminder of how they need to answer the phone. But that is not what you asked the first time around. If you want to change the ringing tone *at the far end* depending on who is calling, that's different. That is just ordinary distinctive ringing, and Asterisk most certainly supports it (in fact, even analogue phones on an FXS card can be given different ringing envelopes; the usual ring-ring, the French-style riing, or even a ring-ring-ring). Your Asterisk at the far end just has to be able to pick up on the caller ID so it knows where the call is coming from, and execute a SipAddHeader() statement to set the appropriate ringing tone. How would I go about setting up telling the phone to change the ring tone in the SIP header? Read the documentation for your phones. For Digium D40s, look here: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=21463877#XMLConfiguration1.1.x- RingtonesElement -- AJS Answers come *after* questions. -- _ -- 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] Set ringtone by dialed number
You can use anything that is discriminable to connect a call to its proper endpoint. SIP headers have the disadvantage that the format may depend on a the phone model. jg -- _ -- 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] Set ringtone by dialed number
A J Stiles wrote: On Tuesday 23 July 2013, Josh Hopkins wrote: The reason for this is we have one primary company office but there are two entities and if someone call the Denver number it would be for 1 organization and would ring differently helping our staff remember how to answer the phone for the Denver organization rather than for the Colorado Springs number which is a different entity. A lot of times people are rushing to answer the phone and do not look at the callerID this would give them and auditory reminder of how they need to answer the phone. But that is not what you asked the first time around. Well, not exactly. He asked how to change the ring TONE If you want to change the ringing tone *at the far end* More properly called ringBACK for those who are knowledgeable in telephony. depending on who is calling, that's different. That is just ordinary distinctive ringing, and Asterisk most certainly supports it (in fact, even analogue phones on an FXS card can be given different ringing envelopes; the usual ring-ring, the French-style riing, or even a ring-ring-ring). But, does Asterisk support distinctive ringing on SIP phones? Isn't that governed within the SIP phone itself Let's not confuse different ringing patterns on analog circuits with SIP. John Novack Your Asterisk at the far end just has to be able to pick up on the caller ID so it knows where the call is coming from, and execute a SipAddHeader() statement to set the appropriate ringing tone. How would I go about setting up telling the phone to change the ring tone in the SIP header? Read the documentation for your phones. For Digium D40s, look here: https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=21463877#XMLConfiguration1.1.x- RingtonesElement -- Dog is my Co-pilot -- _ -- 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 1.8 Service: -r does not give CLI
hello, you should provide more information, like: what have you tried yet? did you restart asterisk? did you find something in the log? greetings On 07/23/2013 04:06 PM, Carlos Rojas wrote: Not it didn't, Did you execute asterisk -r or /usr/sbin/asterisk -r ? If not working did you execute asterisk -gc ? Kind Regards On Mon, Jul 22, 2013 at 10:41 AM, Meadows Hoa meadows_...@yahoo.com mailto:meadows_...@yahoo.com wrote: We have Asterisk1.8.11 and can not move to a newer version right now. But when we run Asterisk as a service, the -r option does not result in giving the CLI prompt? Did the option to get the CLI change? -- _ -- 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] Queue - how to jump to next member after NO ANSWER?
Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again. I know that this is because SIP/100 is still available in the Queue but is it any way to make a Queue witch strategy: call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after given number of time - hump to the next member? Thanks in advance. Jarek -- _ -- 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] Queue - how to jump to next member after NO ANSWER?
Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jarek Jarzebowski Sent: Tuesday, July 23, 2013 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue - how to jump to next member after NO ANSWER? Hi all, I have a Queue with 3 members: SIP/100 SIP/200 SIP/300 When call arrives SIP/100 is ringing.. After given timeout ringing stops but call is not routed to next member but SIP/100 starts ringing again. I know that this is because SIP/100 is still available in the Queue but is it any way to make a Queue witch strategy: call SIP/100 - if it is BUSY, UNAVAILABLE, PAUSED or _NO_ANSWERED_ after given number of time - hump to the next member? Thanks in advance. Jarek -- _ -- 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
