[asterisk-users] multi tenant
Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] multi tenant
Not possible to have same sip usernames. However you can create custA_user1 == 101 custB_user1 == 101 In the dialplan context. Mitul On Oct 30, 2012 12:47 PM, Darin Iv adari...@gmail.com wrote: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] multi tenant
Yes, you can do this. You should point the trunks to the right context and done. Op 30-10-2012 8:15, Darin Iv schreef: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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] multi tenant
Its depends on your incoming trunks. You can define different context to different trunks and your DID/extension will be called as per dialplan in that parituclar context of trunk. On Tue, Oct 30, 2012 at 12:57 PM, Henk Dick h...@osocoms.com wrote: Yes, you can do this. You should point the trunks to the right context and done. Op 30-10-2012 8:15, Darin Iv schreef: Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company Extensions: 101,102,103,104 etc. Company D: Context Company_D IVR Company D Extensions: 101,102,103,104 etc. -- _ -- 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 -- Bharat Lalcheta -- _ -- 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] Cant we have same extension in different context?
Cant we have same extension in different context? This is what we we want in same pbx server? Company A Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B Context Company_B IVR Company B Extensions: 101,102,103,104 etc. -- _ -- 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] Cant we have same extension in different context?
FYI SIP usernames =! Extensions You have to use unique sip usernames to be identified inside dialplan for mapping to extensions [contextA] Exten = 101,1,Dail(sip/custA_user1) [contextB] Exten = 101,1,Dial(sip/custB_user2) Hope this makes it clear. Mitul On Oct 30, 2012 1:16 PM, Darin Iv adari...@gmail.com wrote: Cant we have same extension in different context? This is what we we want in same pbx server? Company A Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B Context Company_B IVR Company B Extensions: 101,102,103,104 etc. -- _ -- 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] Read sometimes disconnects user
But even then, it should do another attempt, play the file again and accept DMTF input, as I specified that there should be 2 attempts. *same = n,Read(mobileNumber,app/input-mobile,10,,2,15)* * * 2012/10/30 Steve Edwards asterisk@sedwards.com On Tue, 30 Oct 2012, Thomas Thomas wrote: I am asking the user to enter his mobile phone followed by # using Read(). From time to time the Read() application disconnects the user while he is typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. It has been my casual observation that the speed at which I enter digits on my phone is unrelated to the speed at which my cell provider delivers the digits to my Asterisk box. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://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] Read sometimes disconnects user
My last message was wrong : If the user doesn't enter any digits within 15 seconds, the file is played again and a second attempt is made. So the timeout parameter specified for Read() works. But if the user enters the numbers, he might get disconnected for some obscure reason. If he enters 04755 and takes too long before entering the following number (timeout digit being set to 2 seconds), then he should anyway take 04755 as the number and continue the script. It is definitily very strange that the user gets disconnected. And that might happen when he entered his 5 first digits, 9 digits, ... seems random. 2012/10/30 Thomas Thomas debussy...@gmail.com But even then, it should do another attempt, play the file again and accept DMTF input, as I specified that there should be 2 attempts. *same = n,Read(mobileNumber,app/input-mobile,10,,2,15)* * * 2012/10/30 Steve Edwards asterisk@sedwards.com On Tue, 30 Oct 2012, Thomas Thomas wrote: I am asking the user to enter his mobile phone followed by # using Read(). From time to time the Read() application disconnects the user while he is typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. It has been my casual observation that the speed at which I enter digits on my phone is unrelated to the speed at which my cell provider delivers the digits to my Asterisk box. -- Thanks in advance, --**--** - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- __**__**_ -- 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-**usershttp://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] Asterisk does not re-register as a sip client after a sip reload if sip.conf or users.conf is changed
Hi all, I have a confusing problem with sip reload. I have an outgoing registration in my sip.conf in the form of register= , and generally when ever I do a sip reload , asterisk sends out a new registration and re-registers. But if I change anything in sip.conf or users.conf , the re-registration does not work anymore after a sip reload (even if I don't change the files and just re-save them). in this case after a sip reload when I issue sip show registry I see 0 registrations and not even a pending reg. I need to add /delete users and extensions on a frequently basis and don't want to lose my registrations to outside or my active calls , and I suppose a sip reload after changing the configs is the only way . I am really confused why it is not working. I am sure that asterisk is re-reading the new config files --new users are added-- but registration is not triggered again. I am not sure if it is a misconfiguration on my side or some kind of bug in asterisk but any help on this issue would be really appreciated. * I tried including an external config file in sip.conf or users.conf but in this case even without changing anything, after a sip reload , I lost my registration. * I tried mysql real time module but since I am working with some websocket clients I have some issues to forward calls from udp clients to websocket ones. *I am working with trunk asterisk 11 (r 373330 ) and I tried it with the latest trunk and same results . Regards Hadi Ams -- _ -- 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 does not re-register as a sip client after a sip reload if sip.conf or users.conf is changed
Hadi Ams wrote: Hi all, Hola, I have a confusing problem with sip reload. I have an outgoing registration in my sip.conf in the form of register= , and generally when ever I do a sip reload , asterisk sends out a new registration and re-registers. But if I change anything in sip.conf or users.conf , the re-registration does not work anymore after a sip reload (even if I don't change the files and just re-save them). in this case after a sip reload when I issue sip show registry I see 0 registrations and not even a pending reg. I need to add /delete users and extensions on a frequently basis and don't want to lose my registrations to outside or my active calls , and I suppose a sip reload after changing the configs is the only way . I am really confused why it is not working. I am sure that asterisk is re-reading the new config files --new users are added-- but registration is not triggered again. I am not sure if it is a misconfiguration on my side or some kind of bug in asterisk but any help on this issue would be really appreciated. Unfortunately this appears to be an issue with Asterisk 11. You can follow progress on solving it at https://issues.asterisk.org/jira/browse/ASTERISK-20611 Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scheduled Maintenance for Asterisk Project community services
On Thursday, November 1st, 2012, the Asterisk community services listed below will be undergoing maintenance (software upgrades and updates). The services will be shut down at approximately 9:00 PM CST (3:00 AM November 2nd UTC), and will return no later than 10:00 PM CST. We apologize in advance for any inconvenience this may cause. The affected services are: issues.asterisk.org/jira -- _ -- 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] Read sometimes disconnects user
Several factors could be affecting the Read() operation. I had problems with 1.4 and DAHDI. They magically disappeared when I went to 10.0. Also, depending on the carrier, you might need to set LONG DTMF tones. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Thomas Sent: Tuesday, October 30, 2012 4:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Read sometimes disconnects user My last message was wrong : If the user doesn't enter any digits within 15 seconds, the file is played again and a second attempt is made. So the timeout parameter specified for Read() works. But if the user enters the numbers, he might get disconnected for some obscure reason. If he enters 04755 and takes too long before entering the following number (timeout digit being set to 2 seconds), then he should anyway take 04755 as the number and continue the script. It is definitily very strange that the user gets disconnected. And that might happen when he entered his 5 first digits, 9 digits, ... seems random. 2012/10/30 Thomas Thomas debussy...@gmail.com But even then, it should do another attempt, play the file again and accept DMTF input, as I specified that there should be 2 attempts. same = n,Read(mobileNumber,app/input-mobile,10,,2,15) 2012/10/30 Steve Edwards asterisk@sedwards.com On Tue, 30 Oct 2012, Thomas Thomas wrote: I am asking the user to enter his mobile phone followed by # using Read(). From time to time the Read() application disconnects the user while he is typing his number, though there is a 15 seconds timeout, and even if I type the number very fast it still may happen to me. It has been my casual observation that the speed at which I enter digits on my phone is unrelated to the speed at which my cell provider delivers the digits to my Asterisk box. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 tel:%2B1-760-468-3867 PST Newline Fax: +1-760-731-3000 tel:%2B1-760-731-3000 -- _ -- 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] Bypass queue wrapup time
Mitch Claborn mitch...@claborn.net writes: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? Slightly hacky solution which only works for ringall: Designate a phone to be in the queue but never get answered. When you are ready for a call early, do a directed pickup of that phone. For a less hacky solution, see https://reviewboard.asterisk.org/r/1619/ /Benny -- _ -- 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 1.6.0 disable cdr account logs?
JR Richardson wrote:
My bad. I sent Igor to the boneyard to fetch 1.6.0.28 and it appears to me
that by commenting out lines 309-312 and doing a fresh make you eliminate
the extra files (or make them empty).
Appriciate the suggestion but commenting out 309-312 refused to compile:
cdr_csv.c
/* if (!ast_strlen_zero(cdr-accountcode)) {
if (writefile(buf, cdr-accountcode))
ast_log(LOG_WARNING, Unable to write CSV
record to account file '%s' : %s\n, cdr-a$
*/ }
You need to place the */ after the } or else they are mismatched and
like you have seen, the universe will explode.
Got it, after properly commenting out that section and re-compiling
and reloading cdr_csv.so, I still get the individual account code
CDR's.
Any other suggestions?
Thanks.
JR
--
JR Richardson
Engineering for the Masses
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Re: [asterisk-users] multi tenant
On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote:
Hi all,
I need to configure DIDs for different companies and they should reach on
different extension with different context. Cant we have same extension
in different context?
This is what we we want
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Context Company_B
IVR Company B
Extensions: 101,102,103,104 etc.
There are multiple ways to do this. One way is the Local dial. We have
done this for companies who are different entities but want to do 3-digit
dial.
Dial(Local/101@company_a#extensions,25)
Where we assume you have a context like:
[company_a#extensions]
exten = 101,1,Dial(SIP/company_a.${EXTEN},25)
Another way is to simply do an include for the other company's extension
context. However that requires that you not duplicate the extension
numbers between the contexts/companies.
--
Carlos Alvarez
TelEvolve
602-889-3003
--
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Re: [asterisk-users] multi tenant
I am attempting to send this again. The mail processor is interpreting the
Asterisk commands in my message as mail processor command and bouncing the
message. That's why where is junk before many of the lines below.
On Tue, Oct 30, 2012 at 12:15 AM, Darin Iv adari...@gmail.com wrote:
Hi all,
I need to configure DIDs for different companies and they should reach on
different extension with different context. Cant we have same extension
in different context?
This is what we we want
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Context Company_B
IVR Company B
Extensions: 101,102,103,104 etc.
There are multiple ways to do this. One way is the Local dial. We have
done this for companies who are different entities but want to do 3-digit
dial.
... Dial(Local/101@company_a#extensions,25)
Where we assume you have a context like:
... [company_a#extensions]
... exten = 101,1,Dial(SIP/company_a.${EXTEN},25)
Another way is to simply do an include for the other company's extension
context. However that requires that you not duplicate the extension
numbers between the contexts/companies.
--
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[asterisk-users] Asterisk 11.0.0 Now Available!
The Asterisk Development Team is pleased to announce the release of Asterisk 11.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases Asterisk 11 is the next major release series of Asterisk. It is a Long Term Support (LTS) release, similar to Asterisk 1.8. For more information about support time lines for Asterisk releases, see the Asterisk versions page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions For important information regarding upgrading to Asterisk 11, please see the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 A short list of new features includes: * A new channel driver named chan_motif has been added which provides support for Google Talk and Jingle in a single channel driver. This new channel driver includes support for both audio and video, RFC2833 DTMF, all codecs supported by Asterisk, hold, unhold, and ringing notification. It is also compliant with the current Jingle specification, current Google Jingle specification, and the original Google Talk protocol. * Support for the WebSocket transport for chan_sip. * SIP peers can now be configured to support negotiation of ICE candidates. * The app_page application now no longer depends on DAHDI or app_meetme. It has been re-architected to use app_confbridge internally. * Hangup handlers can be attached to channels using the CHANNEL() function. Hangup handlers will run when the channel is hung up similar to the h extension; however, unlike an h extension, a hangup handler is associated with the actual channel and will execute anytime that channel is hung up, regardless of where it is in the dialplan. * Added pre-dial handlers for the Dial and Follow-Me applications. Pre-dial allows you to execute a dialplan subroutine on a channel before a call is placed but after the application performing a dial action is invoked. This means that the handlers are executed after the creation of the callee channels, but before any actions have been taken to actually dial the callee channels. * Log messages can now be easily associated with a certain call by looking at a new unique identifier, Call Id. Call ids are attached to log messages for just about any case where it can be determined that the message is related to a particular call. * Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in Asterisk. Unlike traditional ACLs defined in specific module configuration files, Named ACLs can be shared across multiple modules. * The Hangup Cause family of functions and dialplan applications allow for inspection of the hangup cause codes for each channel involved in a call. This allows a dialplan writer to determine, for each channel, who hung up and for what reason(s). * Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE() lets you set some of the configuration options from the general section of features.conf on a per-channel basis. FEATUREMAP() lets you customize the key sequence used to activate built-in features, such as blindxfer, and automon. * Support for DTLS-SRTP in chan_sip. * Support for named pickupgroups/callgroups, allowing any number of pickupgroups and callgroups to be defined for several channel drivers. * IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework. More information about the new features can be found on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Documentation A full list of all new features can also be found in the CHANGES file. http://svnview.digium.com/svn/asterisk/branches/11/CHANGES For a full list of changes in the current release, please see the ChangeLog. http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.0.0 Thank you for your continued support of Asterisk! -- _ -- 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] Stopping dahdi: Unloading DAHDI hardware modules: ERROR: Module dahdi is in error
On Tue, Oct 30, 2012 at 03:17:01AM -0500, Jared Baxley wrote: This issue has been ongoing for some months.. but never really bothered me. However Now I'm experiencing random system lockups which require a hard reset to bring the system back up. It maybe worth running some integrity checks. You could have a bad disk or memory. PRI card is a Sangoma B601 with HEC ... of which I have another using without this issue. I'm not sure in what way Sangoma modifies DAHDI but from the oops it looks like maybe the card's base driver is unregistering incorrectly on a shutdown, then _process_masterspan is running in a broken context. I think your best bet here would be to contact Sangoma tech support directly. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk does not re-register as a sip client after a sip reload if sip.conf or users.conf is changed
At 05:27 AM 10/30/2012, you wrote: I am not sure if it is a misconfiguration on my side or some kind of bug in asterisk but any help on this issue would be really appreciated. Unfortunately this appears to be an issue with Asterisk 11. You can follow progress on solving it at https://issues.asterisk.org/jira/browse/ASTERISK-20611 Is there a good reason you'd release 11.0 today with this serious a bug still in it? Ira -- _ -- 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 does not re-register as a sip client after a sip reload if sip.conf or users.conf is changed
Ira wrote: At 05:27 AM 10/30/2012, you wrote: I am not sure if it is a misconfiguration on my side or some kind of bug in asterisk but any help on this issue would be really appreciated. Unfortunately this appears to be an issue with Asterisk 11. You can follow progress on solving it at https://issues.asterisk.org/jira/browse/ASTERISK-20611 Is there a good reason you'd release 11.0 today with this serious a bug still in it? Asterisk 11 was made available Thursday of last week before this was known, the announcement was delayed due to AstriCon. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bypass queue wrapup time
2012/10/30 Benny Amorsen benny+use...@amorsen.dk Mitch Claborn mitch...@claborn.net writes: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? Slightly hacky solution which only works for ringall: Designate a phone to be in the queue but never get answered. When you are ready for a call early, do a directed pickup of that phone. Are you sure a call which entered into a queue can be directly picked up ? For a less hacky solution, see https://reviewboard.asterisk.org/r/1619/ /Benny -- _ -- 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] Bypass queue wrapup time
The call can't be picked up from queue, per se. It is pick-able when it rings the extension. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Tuesday, October 30, 2012 4:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bypass queue wrapup time 2012/10/30 Benny Amorsen benny+use...@amorsen.dk Mitch Claborn mitch...@claborn.net writes: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? Slightly hacky solution which only works for ringall: Designate a phone to be in the queue but never get answered. When you are ready for a call early, do a directed pickup of that phone. Are you sure a call which entered into a queue can be directly picked up ? For a less hacky solution, see https://reviewboard.asterisk.org/r/1619/ /Benny -- _ -- 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] Bypass queue wrapup time
2012/10/30 Danny Nicholas da...@debsinc.com The call can’t be picked up from queue, per se. It is pick-able when it rings the extension. That's the point : to me, casual @pickupmark mechanism don't work with calls that entered into a queue : the extension rings but you can't pick the call up with a directed pickup. (For general pickup, that's another strory). (and I would be very pleased to be wrong) ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier *Sent:* Tuesday, October 30, 2012 4:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Bypass queue wrapup time ** ** ** ** 2012/10/30 Benny Amorsen benny+use...@amorsen.dk Mitch Claborn mitch...@claborn.net writes: In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? Slightly hacky solution which only works for ringall: Designate a phone to be in the queue but never get answered. When you are ready for a call early, do a directed pickup of that phone. Are you sure a call which entered into a queue can be directly picked up ? For a less hacky solution, see https://reviewboard.asterisk.org/r/1619/ /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bypass queue wrapup time
Olivier oza_4...@yahoo.fr writes: That's the point : to me, casual @pickupmark mechanism don't work with calls that entered into a queue : the extension rings but you can't pick the call up with a directed pickup. (For general pickup, that's another strory). (and I would be very pleased to be wrong) That seems to be fixed a long time ago, if I read the various issues correctly. I haven't actually tried it. /Benny -- _ -- 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
