Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk Call Manager/1.1 Action: Login Username: allpage Secret: xxxxxxxxxxxxxxxxxxxxxxx Events: off Action: Originate Channel: SIP/7527 Context: all-page Priority: 1 Variable: SIPADDHEADER="Call-Info: sip:172.30.254.211" Variable: ALERT_INFO="Ring Answer" Extension: CallerID: System Page Action: Logoff Here my phone SIP/7527 is ringing but not auto answer. why ? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110314/2a87d133/attachment.htm>
On 03/14/2011 10:01 AM, satish patel wrote:> Hey Guys, > > I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi > stopped working look like asterisk 1.8 did some changes in manager apps > i am doing following.. my phone is ringing but not auto answer could you > give me some issue what i am doing wrong ?The manager interface has indeed changed between 1.2 and 1.8 (likely it has changed many times), and you would do yourself a world of good to read through the upgrade notes that came with Asterisk 1.8 to understand how you might need to change your scripts. In addition, Asterisk 1.8 has a built-in Page() application you can use from the dialplan to achieve what it appears you were trying to achieve with your AGI script. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org
If I was worried I'd record the 'page' first - and then play it back to 50 handsets at a time (using a loop). -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel Sent: 14 March 2011 16:25 To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom Thanks Kevin, I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Just worried about my asterisk. I don't want to crach :( -Satish> Date: Mon, 14 Mar 2011 11:18:36 -0500 > From: kpfleming at digium.com > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > On 03/14/2011 10:01 AM, satish patel wrote: > > Hey Guys, > > > > I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi > > stopped working look like asterisk 1.8 did some changes in managerapps> > i am doing following.. my phone is ringing but not auto answer couldyou> > give me some issue what i am doing wrong ? > > The manager interface has indeed changed between 1.2 and 1.8 (likelyit> has changed many times), and you would do yourself a world of good to > read through the upgrade notes that came with Asterisk 1.8 tounderstand> how you might need to change your scripts. > > In addition, Asterisk 1.8 has a built-in Page() application you canuse> from the dialplan to achieve what it appears you were trying toachieve> with your AGI script. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype:kpfleming> 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-usersIf you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085.
...http://ofps.oreilly.com/titles/9780596517342/ch11.html if you're not sure on Multicast (near the bottom). -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven Howes Sent: 14 March 2011 16:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom On 14 Mar 2011, at 16:24, satish patel wrote: I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Do they support multicast? S If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085.
Oops - from the very bottom of that page (no pun intended) : "So far as we can tell, Polycom sets do not support multicast. We certainly were not able to find a way to use it." -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven Howes Sent: 14 March 2011 16:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom On 14 Mar 2011, at 16:24, satish patel wrote: I test page application and it works but i am worried about i have 200 SIP phone. Do you think asterisk page application can handle that number of page ? Do they support multicast? S If you have received this communication in error we would appreciate you advising us either by telephone or return of e-mail. The contents of this message, and any attachments, are the property of DataVox, and are intended for the confidential use of the named recipient only. If you are not the intended recipient, employee or agent responsible for delivery of this message to the intended recipient, take note that any dissemination, distribution or copying of this communication and its attachments is strictly prohibited, and may be subject to civil or criminal action for which you may be liable. Every effort has been made to ensure that this e-mail or any attachments are free from viruses. While the company has taken every reasonable precaution to minimise this risk, neither company, nor the sender can accept liability for any damage which you sustain as a result of viruses. It is recommended that you should carry out your own virus checks before opening any attachments. Registered in England. No. 27459085.
[default] exten => 777,1,Answer() exten => 777,n,Record(/var/lib/asterisk/sounds/page:gsm) exten => 777,n,Originate(Local/pb at dv-ip,exten,page-it,s,1) exten => 777,n,Hangup() exten => pb,1,Answer() exten => pb,n,Playback(page) [page-it] exten => s,1,Set(page1=SIP/801&SIP/802&SIP/803) ; etc etc exten => s,n,SIPAddHeader(Call-Info: \;answer-after=0) exten => s,n,SIPAddHeader(Answer-Mode: Auto) exten => s,n,SIPAddHeader(P-Auto-answer: normal) exten => s,n,SIPAddHeader(Alert-Info: info=alert-autoanswer) exten => s,n,Page(${page1}) This works for me with a Yealink T28, a Linksys SPA-941, an Aastre 6755i and a Grandstream BT-200. Paging person dials 777 and records msg. Msg is then played to other handsets when # is pressed. Remember, the person paging can't hangup until the page has been played (in this example). HTH -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel Sent: 15 March 2011 15:17 To: asterisk-users Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom Hey, Could you give me some idea how to do this ? I meant record and play ? do you want me to use .call file ? -Satish> Date: Mon, 14 Mar 2011 16:29:19 +0000 > From: andy at datavox.co.uk > To: asterisk-users at lists.digium.com > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > If I was worried I'd record the 'page' first - and then play it backto> 50 handsets at a time (using a loop). > > > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish > patel > Sent: 14 March 2011 16:25 > To: asterisk-users > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > > Thanks Kevin, > > I test page application and it works but i am worried about i have 200 > SIP phone. Do you think asterisk page application can handle thatnumber> of page ? > > Just worried about my asterisk. I don't want to crach :( > > -Satish > > > > > Date: Mon, 14 Mar 2011 11:18:36 -0500 > > From: kpfleming at digium.com > > To: asterisk-users at lists.digium.com > > Subject: Re: [asterisk-users] Asterisk 1.8 paging with ploycom > > > > On 03/14/2011 10:01 AM, satish patel wrote: > > > Hey Guys, > > > > > > I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi > > > stopped working look like asterisk 1.8 did some changes in manager > apps > > > i am doing following.. my phone is ringing but not auto answercould> you > > > give me some issue what i am doing wrong ? > > > > The manager interface has indeed changed between 1.2 and 1.8 (likely > it > > has changed many times), and you would do yourself a world of goodto> > read through the upgrade notes that came with Asterisk 1.8 to > understand > > how you might need to change your scripts. > > > > In addition, Asterisk 1.8 has a built-in Page() application you can > use > > from the dialplan to achieve what it appears you were trying to > achieve > > with your AGI script. > > > > -- > > Kevin P. Fleming > > Digium, Inc. | Director of Software Technologies > > Jabber: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: > kpfleming > > 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 everyThurs:> > http://www.asterisk.org/hello > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > If you have received this communication in error we would appreciate > you advising us either by telephone or return of e-mail. The contents > of this message, and any attachments, are the property of DataVox, > and are intended for the confidential use of the named recipient only. > If you are not the intended recipient, employee or agent responsible > for delivery of this message to the intended recipient, take note that > any dissemination, distribution or copying of this communication and > its attachments is strictly prohibited, and may be subject to civil or > criminal action for which you may be liable. > Every effort has been made to ensure that this e-mail or anyattachments> are free from viruses. While the company has taken every reasonable > precaution to minimise this risk, neither company, nor the sender can > accept liability for any damage which you sustain as a result ofviruses.> It is recommended that you should carry out your own virus checks > before opening any attachments. > > Registered in England. No. 27459085. > > > > -- > _____________________________________________________________________ > -- 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