Re: [asterisk-users] Page() bumps user out of a call
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a call,
or
2. use some other function that will not bump a call?
Has anyone else run into this?
Thanks
Cassius
Here is my intercom context:
[intercom]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
exten = t,1,NoOp(timeout)
exten = t,n,Playback(sorry-youre-having-problemsgoodbye)
exten = t,n,Hangup()
exten = *,1,SIPAddHeader(Call-Info:sip:${SERVER_IP}\;answer-after=0)
exten = *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here
exten = _,1,SIPAddHeader(Call-Info:
sip:${SERVER_IP}\;answer-after=0) ; 4 digit extensions
exten = _,n,Dial(SIP/${EXTEN})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk
in now? You might want to checkout the DEVICE_STATE() function. Should
be able to build your ALL-PAGE-EXTS while leaving out the busy
extensions. Probably not the best solution, but the first one I thought of.
--
Russ Meyerriecks
Digium | Linux Kernel Developer
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Re: [asterisk-users] Page() bumps user out of a call
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a
call,
or
2. use some other function that will not bump a call?
Has anyone else run into this?
Thanks
Cassius
Here is my intercom context:
[intercom]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
exten = t,1,NoOp(timeout)
exten = t,n,Playback(sorry-youre-having-problemsgoodbye)
exten = t,n,Hangup()
exten = *,1,SIPAddHeader(Call-Info:sip:${SERVER_IP}\;answer-after=0)
exten = *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here
exten = _,1,SIPAddHeader(Call-Info:
sip:${SERVER_IP}\;answer-after=0) ; 4 digit extensions
exten = _,n,Dial(SIP/${EXTEN})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk in
now? You might want to checkout the DEVICE_STATE() function. Should be
able to build your ALL-PAGE-EXTS while leaving out the busy extensions.
Probably not the best solution, but the first one I thought of.
This may be a better solution, actually. Checkout example 1. It sets up
a macro to handle the check for each extension.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
--
Russ Meyerriecks
Digium | Linux Kernel Developer
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Re: [asterisk-users] Page() bumps user out of a call
On 6/14/11 4:37 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote:
On 6/14/11 4:25 PM, Russ Meyerriecks wrote:
On 6/14/11 9:26 AM, Cassius Smith wrote:
Hello all,
I'm having a problem with my intercom function that I use for
under-chin
paging. I'm running 1.6.2.13 on this server, and we use Linksys
SPA-942's
for our general phones. I have a global defined which has all the SIP
channels concatenated together - this is ${ALL-PAGE-EXTS}.
The problem comes when a user is on the line, and someone else uses the
intercom function to page all extensions, the call in progress gets
disconnected. I'm wondering if there is a way to either:
1. dynamically figure out the subset of extensions that are not in a
call,
or
2. use some other function that will not bump a call?
Has anyone else run into this?
Thanks
Cassius
Here is my intercom context:
[intercom]
exten = s,1,Answer
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
exten = t,1,NoOp(timeout)
exten = t,n,Playback(sorry-youre-having-problemsgoodbye)
exten = t,n,Hangup()
exten = *,1,SIPAddHeader(Call-Info:sip:${SERVER_IP}\;answer-after=0)
exten = *,n,Page(${ALL-PAGE-EXTS}) ; add all your devices here
exten = _,1,SIPAddHeader(Call-Info:
sip:${SERVER_IP}\;answer-after=0) ; 4 digit extensions
exten = _,n,Dial(SIP/${EXTEN})
Hey Cassius!
Nice to hear from you, what crazy country are you deploying Asterisk in
now? You might want to checkout the DEVICE_STATE() function. Should be
able to build your ALL-PAGE-EXTS while leaving out the busy extensions.
Probably not the best solution, but the first one I thought of.
This may be a better solution, actually. Checkout example 1. It sets up
a macro to handle the check for each extension.
http://www.voip-info.org/wiki/view/Asterisk+cmd+Page
Hi Russ,
Thanks for this. I was thinking of the DEVICE_STATE() also, just hoping
someone
Had a snippet that might make it easier. I've implemented something very
much like
The example 1 code on the referenced page. (The above code was actually
from example 2!).
I will have the crew in Vienna check it out when they get into the office.
Cassius
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Re: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
That's almost certainly your problem. When you run sidecars with the Polycom 601, you can't rely on PoE - there isn't enough power supplied. Connect your powerpack to the phone and the problem should go away. Semi random reboots are not uncommon on the 601 with sidecars if you're running it on PoE. Well, I wish it were that easy. Really, I do!!! I put a 601 power supply on Friday afternoon. Have have had 2 reboots already this morning during pages. The 601 simply can't handle the traffic of 23 simultaneous Buddy Watch updates. If a call comes in during a page. It will crash every time. We're getting a 650 in to see if that will fix the problem (as it did for others) Thanks Bill ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. That's almost certainly your problem. When you run sidecars with the Polycom 601, you can't rely on PoE - there isn't enough power supplied. Connect your powerpack to the phone and the problem /should/ go away. Semi random reboots are not uncommon on the 601 with sidecars if you're running it on PoE. That makes sense but in my case the 601 w/3 sidecars did not reboot at all and it is run from POE. The 650 just seems to perform much better. JR --- JR Richardson Engineering for the Masses ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
JR Richardson Engineering for the Masses -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of asterisk-users- [EMAIL PROTECTED] Sent: Saturday, March 01, 2008 12:00 PM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 44, Issue 1 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] Oh yes! This has been killing us for about a year. We've had several conference calls with my phone vendor and Polycom and it's still not fixed (or even determined why it is happening). Polycom keeps saying, upgrade to the next version of the firmware. We upgrade, still a problem. (again, for over a year!) In my case, the Polycom 601 actually reboots when we page! When it comes back up, I have a phantom meetme on the Asterisk system and none of the sidecar lights are correct. Sometimes, they simply stop updating completely. Just FYI, go to the CLI and type meetme. You'll get the conference ID and the number of users. Then, type meetme kick confID 01 Using, of course, the conference ID. The 01 is the user that initiated the meetme. So, when you kick 01, the rest go away politely! This keeps us from having to restart Asterisk. We are on Bootrom 3.2.3.0002 and SIP 2.2.0.0047 as of yesterday and we STILL have the problem. Our setup is one Polycom 601 and 25 Polycom 501s that are being paged. The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. Our next plan is to get a 650 and see if it can handle the traffic. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, February 29, 2008 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses We sent a Polycom 650 on sight and replaced the 601. Paging works fine now and that extension has not dropped off the network at all, this customer group pages a lot, probably 20+ times a day. The previous condition with the Polycom 601 is not present with the 650. We made no changes to Asterisk or phone configuration. Both phones were running 2.1.1. Hope this helps. JR --- JR Richardson Engineering for the Masses ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue WORKING NOW
The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. That's almost certainly your problem. When you run sidecars with the Polycom 601, you can't rely on PoE - there isn't enough power supplied. Connect your powerpack to the phone and the problem /should/ go away. Semi random reboots are not uncommon on the 601 with sidecars if you're running it on PoE. ___ -- 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] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue
Oh yes! This has been killing us for about a year. We've had several conference calls with my phone vendor and Polycom and it's still not fixed (or even determined why it is happening). Polycom keeps saying, upgrade to the next version of the firmware. We upgrade, still a problem. (again, for over a year!) In my case, the Polycom 601 actually reboots when we page! When it comes back up, I have a phantom meetme on the Asterisk system and none of the sidecar lights are correct. Sometimes, they simply stop updating completely. Just FYI, go to the CLI and type meetme. You'll get the conference ID and the number of users. Then, type meetme kick confID 01 Using, of course, the conference ID. The 01 is the user that initiated the meetme. So, when you kick 01, the rest go away politely! This keeps us from having to restart Asterisk. We are on Bootrom 3.2.3.0002 and SIP 2.2.0.0047 as of yesterday and we STILL have the problem. Our setup is one Polycom 601 and 25 Polycom 501s that are being paged. The 601 is powered by PoE with 2 sidecars, so Polycom wants us to put an actual Power Supply on the phone - thinking the voltage is dropping and causing the reboot. I don't buy that, but we are putting one on next Monday. We'll see. Our next plan is to get a 650 and see if it can handle the traffic. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Friday, February 29, 2008 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Page app, Polycom IP 601, 60 SIP peers, Interesting Issue Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the receptionist hangs up the page, the BYE doesn't get back to Asterisk to release all the Page channels so they stay open. I have to restart Asterisk to release all the open SIP Channels. What I think is happening is when all the SIP peers are paged, Asterisk sends 60 hint notifications to the IP 601 and the phone is overloaded and can't respond to SIP POKE or process the BYE message back to Asterisk properly. I'm wondering if I upgrade to a new IP 650 with a faster processor, will this eliminate the issue? Has anyone experienced this or have ideas for resolution or further troubleshooting? Thanks. JR -- JR Richardson Engineering for the Masses ___ -- 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] Page Command
At the current moment there is no way. You would need to specify all phones. If you were using real time you can write an agi that would fetch a list of all phones and then page them all. - Original Message - From: Anciso, Roy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, November 17, 2007 9:05 PM Subject: [asterisk-users] Page Command Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [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___ --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] Page Command
At 23:16 11/19/2007, Dovid B wrote: At the current moment there is no way. You would need to specify all phones. If you were using real time you can write an agi that would fetch a list of all phones and then page them all. There seems to be one here: http://www.voip-info.org/wiki/view/Polycom+auto-answer+config - Original Message - From: mailto:[EMAIL PROTECTED]Anciso, Roy To: mailto:asterisk-users@lists.digium.comAsterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, November 17, 2007 9:05 PM Subject: [asterisk-users] Page Command Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 mailto:[EMAIL PROTECTED][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 ___ --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] Page + ParkAndAnnounce
On Friday 15 December 2006 4:18 am, Apesys wrote: exten = s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE(LOCAL/[EMAIL PROTECTED] o pageLOCAL/[EMAIL PROTECTED]|) why not Local/[EMAIL PROTECTED], and then have something like this: [group_page] exten = ,1,Dial(SIP/555) exten = ,1,Dial(SIP/123SIP/456SIP/789) exten = ,1,Dial(SIP/123SIP/789) ... is that closer to what you're looking for? -A. ___ --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] Page() Function Timeout
On 11/16/06 06:06 David Gagnon said the following: Which version are you using? There was a problem in 1.2.12.1 with the page application. Update to 1.2.13. what was the problem ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] Page() Function Timeout
BAH!
My Makefile in the apps folder was missing
app_page.c. I added it, recompiled, page is working
properly.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
WilliamsSent: Wednesday, November 15, 2006 10:33 AMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject:
[asterisk-users] Page() Function Timeout
I'm trying to use a
simple page function. It starts a MeetMe conference with the devices I've
listed, but the devices hang up after 3-5 seconds. After doing some
research I found this was a problem, and I needed to remove a (5) from
app_page.c
Well, my app_page.c
didn't have the (5). I did make clean; make install again just in case I
had some weird compiled version installed that had the (5) in it. After
compiling I restarted the asterisk service and tried paging again and still had
the same problem.
In the CLI I get the
following, which you can see the (5) is still in there somehow.
-- Playing 'beep' (language 'en') -- Launching
MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038 --
Created MeetMe conference 1023 for conference
'1010553064d' -- Launching MeetMe(1010553064d|mqxdw(5)) on
SIP/717-09a48758
I've grep'd the
entire src folder for \(5\) as well as qxd trying to find all instances of this,
and the only ones are listed in the app_page.c file. Any suggestions on
where to get this rogue (5) out of here?
snprintf(meetmeopts,
sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(flags, PAGE_DUPLEX)
? "" : "m");
and
if (!res)
{
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid,
$
pbx_exec(chan, app, meetmeopts,
1); }
are the only
sections of the app_page.c that have the meetme call in it.
My page functions,
fwiw, both have the same problem:
;Paging
exten =
760,1,SIPAddHeader(Call-Info: answer-after=0)exten =
760,2,Page(SIP/717SIP/710SIP/702|d)exten =
760,3,Hangup
exten =
761,1,SIPAddHeader(Call-Info: answer-after=0)exten =
761,2,Page(SIP/717SIP/710SIP/702)exten =
761,3,Hangup
Any suggestions
would be very helpful.
Ken
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Re: [asterisk-users] Page() Function Timeout
Ken Williams wrote:
I'm trying to use a simple page function. It starts a MeetMe
conference with the devices I've listed, but the devices hang up after
3-5 seconds. After doing some research I found this was a problem,
and I needed to remove a (5) from app_page.c
Well, my app_page.c didn't have the (5). I did make clean; make
install again just in case I had some weird compiled version installed
that had the (5) in it. After compiling I restarted the asterisk
service and tried paging again and still had the same problem.
In the CLI I get the following, which you can see the (5) is still in
there somehow.
-- Playing 'beep' (language 'en')
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038
-- Created MeetMe conference 1023 for conference '1010553064d'
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/717-09a48758
I've grep'd the entire src folder for \(5\) as well as qxd trying to
find all instances of this, and the only ones are listed in the
app_page.c file. Any suggestions on where to get this rogue (5) out
of here?
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
and
if (!res) {
snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd,
confid, $
pbx_exec(chan, app, meetmeopts, 1);
}
are the only sections of the app_page.c that have the meetme call in it.
My page functions, fwiw, both have the same problem:
;Paging
exten = 760,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 760,2,Page(SIP/717SIP/710SIP/702|d)
exten = 760,3,Hangup
exten = 761,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 761,2,Page(SIP/717SIP/710SIP/702)
exten = 761,3,Hangup
Any suggestions would be very helpful.
I had the same problem and ended up changing the 5 to a 300. If you
don't specify a (N) after the 'w', I believe it defaults to 5.
Steve
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RE: [asterisk-users] Page() Function Timeout
Which version are you using? There was a problem in 1.2.12.1 with the page
application. Update to 1.2.13.
David
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steven
Ringwald
Envoyé : 15 novembre 2006 13:45
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Page() Function Timeout
Ken Williams wrote:
I'm trying to use a simple page function. It starts a MeetMe
conference with the devices I've listed, but the devices hang up after
3-5 seconds. After doing some research I found this was a problem,
and I needed to remove a (5) from app_page.c
Well, my app_page.c didn't have the (5). I did make clean; make
install again just in case I had some weird compiled version installed
that had the (5) in it. After compiling I restarted the asterisk
service and tried paging again and still had the same problem.
In the CLI I get the following, which you can see the (5) is still in
there somehow.
-- Playing 'beep' (language 'en')
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038
-- Created MeetMe conference 1023 for conference '1010553064d'
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/717-09a48758
I've grep'd the entire src folder for \(5\) as well as qxd trying to
find all instances of this, and the only ones are listed in the
app_page.c file. Any suggestions on where to get this rogue (5) out
of here?
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
and
if (!res) {
snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd,
confid, $
pbx_exec(chan, app, meetmeopts, 1);
}
are the only sections of the app_page.c that have the meetme call in it.
My page functions, fwiw, both have the same problem:
;Paging
exten = 760,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 760,2,Page(SIP/717SIP/710SIP/702|d)
exten = 760,3,Hangup
exten = 761,1,SIPAddHeader(Call-Info: answer-after=0)
exten = 761,2,Page(SIP/717SIP/710SIP/702)
exten = 761,3,Hangup
Any suggestions would be very helpful.
I had the same problem and ended up changing the 5 to a 300. If you
don't specify a (N) after the 'w', I believe it defaults to 5.
Steve
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Re: [asterisk-users] Page system using the sound card
Xue Liangliang wrote: Hi, I followed the steps in the voip-info wiki to implement a page Lets see your dial plan for access and your modules.conf. Are you using OSS or ALSA? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Page hangs up after 5 seconds
OK... A bit more research done... This problem does not occur in version
1.2.7.1, which was the platform where we developed our dialplan.
Looking at a diff between app_page.c for the two version reveals that
the only change that has been done is the addition of (5) to the w option:
1.2.7.1, line 182:
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
1.2.12.1, line 182:
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw(5), confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
Why this change? And I can't imagine that it is the intended behaviour.
Hasn't anyone else noticed this? Or are we doing something fundamentally
wrong?
I still do not understand what the usage and result of the w option are,
could someone elaborate?
// Torbjörn
Torbjörn Abrahamsson wrote:
Hi asterisk-users,
We are using Asterisk 1.2.12.1, and are trying to use the Page
application. It seems to work but after approx 4-5 seconds the call is
hung up.
The dialplan code look like this:
exten = _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2})
exten = _*2XX,n,GotoIf($[ ${PAGING_DEVICES} = invalid ]?i,1)
exten = _*2XX,n,SIPAddHeader(Call-Info: sip:192.168.20.1\; answer-after=0)
exten = _*2XX,n,Page(${PAGING_DEVICES},dq)
The CLI outputs the following:
-- Executing AGI(SIP/snom1-b7d0c328, get-paging-devices.agi|01)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/get-paging-devices.agi
-- AGI Script get-paging-devices.agi completed, returning 0
-- Executing GotoIf(SIP/snom1-b7d0c328, 0?i|1) in new stack
-- Executing SIPAddHeader(SIP/snom1-b7d0c328, Call-Info:
sip:192.168.20.1; answer-after=0) in new stack
-- Executing Page(SIP/snom1-b7d0c328, SIP/snom1SIP/snom3|dq) in
new stack
-- Created MeetMe conference 1023 for conference '2028709590d'
-- Launching MeetMe(2028709590d|qxdw(5)) on SIP/snom3-08984140
-- Hungup 'Zap/pseudo-1436409106'
== Spawn extension (wx3trunk2, *201, 4) exited non-zero on
'SIP/snom1-b7d0c328'
-- Executing Hangup(SIP/snom1-b7d0c328, ) in new stack
The 'full' log has this contents:
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Goto'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
Goto(SIP/snom1-b7d0c328, wx3trunk2|*201|1) in new stack
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Goto (wx3trunk2,*201,1)
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'AGI'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
AGI(SIP/snom1-b7d0c328, get-paging-devices.agi|01) in new stack
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/get-paging-devices.agi
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- AGI Script
get-paging-devices.agi completed, returning 0
Oct 16 11:01:12 DEBUG[6767] pbx.c: Expression result is '0'
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'GotoIf'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
GotoIf(SIP/snom1-b7d0c328, 0?i|1) in new stack
Oct 16 11:01:12 DEBUG[6767] pbx.c: Not taking any branch
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'SIPAddHeader'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
SIPAddHeader(SIP/snom1-b7d0c328, Call-Info: sip:192.168.20.1;
answer-after=0) in new stack
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Page'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
Page(SIP/snom1-b7d0c328, SIP/snom1SIP/snom3|dq) in new stack
Oct 16 11:01:12 DEBUG[6767] chan_sip.c: sip_answer(SIP/snom1-b7d0c328)
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Building dynamic conference
'2028709590d'
Oct 16 11:01:12 DEBUG[6767] chan_zap.c: Using channel -2
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Created MeetMe conference
1023 for conference '2028709590d'
Oct 16 11:01:12 DEBUG[6767] channel.c: Set channel SIP/snom1-b7d0c328 to
write format slin
Oct 16 11:01:12 DEBUG[6767] channel.c: Set channel SIP/snom1-b7d0c328 to
read format slin
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Placed channel
SIP/snom1-b7d0c328 in ZAP conf 1023
Oct 16 11:01:12 DEBUG[6772] app_queue.c: Device 'SIP/snom1' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
Oct 16 11:01:12 DEBUG[6773] app_queue.c: Device 'Zap/pseudo' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Allocating new SIP dialog for
(No Call-ID) - INVITE (With RTP)
Oct 16 11:01:12 DEBUG[6771] res_config_mysql.c: MySQL RealTime:
Everything is fine.
Oct 16 11:01:12 DEBUG[6771] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sipusers WHERE name = 'snom3'
Oct 16 11:01:12 VERBOSE[6771] logger.c: -- SIP Seeding peer from
astdb: 'snom3' at [EMAIL PROTECTED]:59283 for 60
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Allocating new SIP dialog for
(No Call-ID) - OPTIONS (No RTP)
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Setting NAT on RTP to 524288
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Outgoing
Re: [asterisk-users] Page() paging application problem
Michael wrote: extreme echo. After 4 seconds, however, the audio transmission stops. Even though the audio stops, the MeetMe is still in progress until the user who initiated the page hangs up. Maybe the 4 second time limit is within the AGI itself? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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: RE: [asterisk-users] Page() paging application problem
The AGI has already completed, and the Page() application has started. It is not possible for the Page() application to run without the AGI completing its task.-Michael In reply to:Maybe the 4 second time limit is within the AGI itself?Doug ___ --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] Page()
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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Re: [asterisk-users] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0)
exten = s,n,Playback(beep)
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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RE: [asterisk-users] Page()
What is the module I should be loading and how do I load it?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leonardo
Kamache (Gmail)
Sent: Wednesday, August 16, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0) exten =
s,n,Playback(beep) exten = s,n,Set(TIMEOUT(digit)=5) exten =
s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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Re: [asterisk-users] Page()
I just got done implementing this on a Realtime system and it works
flawlessly. You need to create a macro named page that you call from
the dialplan. Please refer to the wiki for more details:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+page
Good luck!
Joe
Dennis P. Clark wrote:
What is the module I should be loading and how do I load it?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leonardo
Kamache (Gmail)
Sent: Wednesday, August 16, 2006 10:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Page()
Hi there;
Did you load the respective module?
Regards;
LK
On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote:
I receive the following error in the Asterisk console when I try to
execute the Page() application:
WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page'
for extention (intercom, *, 1)
EXTENSIONS.CONF
[Default]
Exten = *80,1,Goto(intercom,s,1)
[intercom]
exten = s,1,Answer
exten = s,n,SIPAddHeader(Call-Info: answer-after=0) exten =
s,n,Playback(beep) exten = s,n,Set(TIMEOUT(digit)=5) exten =
s,n,WaitExten(10)
;Page
exten = *,1,Page(SIP/2000x1)
;Intercom
exten = _,1,Dial(SIP/${EXTEN})
Any clues?
Dennis Clark
DENPRO
WRK 207.618.1998
CEL 443.415.0527
FAX 1.888.811.8809
[EMAIL PROTECTED]
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Re: [asterisk-users] Page()
Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Page()
1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Page()
Dennis P. Clark wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug The Page application is app_page.so (located in /usr/lib/asterisk/modules on RH systems). It is present in v1.2.10 and at least at SVN-trunk-r16869M (June 4, 2006). From the CLI, do a 'show modules like page' to see if it is loaded. ___ --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] Page()
in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Page()
I am running Fedora 5 Cat /proc/sys/kernel/osrelease 2.6.15-1.2054_FC5 Zaptel 1.2.7 was not installed Edited xpp_usb.c and wcusb.c files in Zaptel to get it to compile and install by commenting out the following .owner = THIS_MODULE, I receive the following from CLI when I run load module app_page.so WARNING[28359]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_page.so: cannot open shared object file: No such file or directory And yes app_page.so does not exist in /usr/lib/asterisk Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Page()
I guess it might not get compiled if you don't have a timer. Install ztdummy, recompile asterisk and try again. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: I am running Fedora 5 Cat /proc/sys/kernel/osrelease 2.6.15-1.2054_FC5 Zaptel 1.2.7 was not installed Edited xpp_usb.c and wcusb.c files in Zaptel to get it to compile and install by commenting out the following .owner = THIS_MODULE, I receive the following from CLI when I run load module app_page.so WARNING[28359]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_page.so: cannot open shared object file: No such file or directory And yes app_page.so does not exist in /usr/lib/asterisk Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 ___ --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] Page()
That fixed it. Thanks! Here is what I did to fix Asterisk -crv (enter CLI) Stop gracefully (Shutdown Asterisk) Cd /usr/src/asterisk-1.2.10 (Go to unpacked Asterisk installation files) Make install (install asterisk) Asterisk (start asterisk) Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 5:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() I guess it might not get compiled if you don't have a timer. Install ztdummy, recompile asterisk and try again. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: I am running Fedora 5 Cat /proc/sys/kernel/osrelease 2.6.15-1.2054_FC5 Zaptel 1.2.7 was not installed Edited xpp_usb.c and wcusb.c files in Zaptel to get it to compile and install by commenting out the following .owner = THIS_MODULE, I receive the following from CLI when I run load module app_page.so WARNING[28359]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/app_page.so: cannot open shared object file: No such file or directory And yes app_page.so does not exist in /usr/lib/asterisk Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 16, 2006 12:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() in the CLI do: show applications like page if you something there then you have it loaded, otherwise do: load app_page.so if that fails my guess is you need zaptel loaded first. On 8/16/06, Dennis P. Clark [EMAIL PROTECTED] wrote: 1.2.10 Dennis Clark DENPRO WRK 207.618.1998 CEL 443.415.0527 FAX 1.888.811.8809 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Wednesday, August 16, 2006 11:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Page() Dennis P. Clark wrote: I receive the following error in the Asterisk console when I try to execute the Page() application: WARNING[24360]: pbx.c:1700 pbx_extention_helper: No application 'Page' for extention (intercom, *, 1) What version of Asterisk are you running? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 ___ --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] Page Groups
Any phone that supports Auto answer can do this, among those phones: Cisco 796x, Polycom 3xx,430,50x,60x, SPA9xx. The SPA9xx (which support auto answer) will even support it while you are on the phone, it will however put the current conversation on hold for the duration of the page. On 8/15/06, Curt Shaffer [EMAIL PROTECTED] wrote: I have a company that I am going to be moving away from a legacy PBX to Asterisk. They use page zones pretty heavy and I would like to keep that functionality. Basically when someone is not at their desk the receptionist pages all of the phones, telling them there is a call. Does anyone out there know of the best phones to do this with and if it is really even possible. I see that intercom is not supported and paging appears to be minimally supported. Thanks Curt ___ --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] Page Groups
For
intercom, do you mean placing a call that is automatically answered by the
called party?
If so,
the following works for legacy phones connected via a Citel Handset Gateway,
amongst others:
exten
= _*803X.,1,Macro(user-callerid)exten =
_*803X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten
= _*803X.,3,SIPAddHeader(Answer-Mode: Auto) exten =
_*803X.,4,Dial(SIP/${EXTEN:4})
(so
you dial *803 and then the extension number you want to
target)
Similar techniques can be used for page.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Curt
ShafferSent: 15 August 2006 17:16To: 'Asterisk Users
Mailing List - Non-Commercial Discussion'Subject: [asterisk-users]
Page Groups
I have a company that I am going
to be moving away from a legacy PBX to Asterisk. They use page zones pretty
heavy and I would like to keep that functionality. Basically when someone is
not at their desk the receptionist pages all of the phones, telling them there
is a call. Does anyone out there know of the best phones to do this with and
if it is really even possible. I see that intercom is not supported and paging
appears to be minimally supported.
Thanks
Curt
___
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RE: [asterisk-users] Page Groups
For paging, and I have not done this yet,
you would probably have to invite all the phones to a conference with the
auto-answer
The below works great for intercom though .
Polycom which I have used
exten = _*7XXX,1,SetVar(ALERT_INFO=Ring
Answer)
exten = _*7XXX,2,Dial.blah
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Langstaff
Sent: Tuesday, August 15, 2006
12:46 PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Page
Groups
For intercom, do you mean placing a call
that is automatically answered by the called party?
If so, the following works for legacy
phones connected via a Citel Handset Gateway, amongst others:
exten = _*803X.,1,Macro(user-callerid)
exten = _*803X.,2,SetVar(_ALERT_INFO=info=alert-autoanswer)
exten =
_*803X.,3,SIPAddHeader(Answer-Mode: Auto)
exten = _*803X.,4,Dial(SIP/${EXTEN:4})
(so you dial *803 and then the extension
number you want to target)
Similar techniques can be used for page.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Curt Shaffer
Sent: 15 August 2006 17:16
To: 'Asterisk
Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Page
Groups
I have a company that I am going to be moving away from a
legacy PBX to Asterisk. They use page zones pretty heavy and I would like to
keep that functionality. Basically when someone is not at their desk the
receptionist pages all of the phones, telling them there is a call. Does anyone
out there know of the best phones to do this with and if it is really even
possible. I see that intercom is not supported and paging appears to be
minimally supported.
Thanks
Curt
___
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Re: [Asterisk-Users] Page() command and file playback
Thanks for your email, I am currently on annual leave and will return on the 19th July. Many Thanks Scott Pinhorne ___ --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] Page cmd FOP
Hi, On 6/1/06, Mike Clark [EMAIL PROTECTED] wrote: We have a location with around 50 Polycom phones. Asterisk version is 1.2.1 We have implemented paging through the Polycoms, which works great. We are now trying to get FOP .26 going for the receptionist. It seems to work fine, except that when someone does and overhead page, about 3/4 of the phones will continue to show that they are on the phone after the page is complete and hung up. It clears up for any extension when they use that phone. Any ideas? I will need to look at op_server.pl level 1 debug output while doing the page until the problem shows up to see if it is a bug in FOP or not. You can send the capture off list to me together with a description of your problem and a copy of your op_buttons.cfg file. You can continue asking FOP related questions in its mailing list, you can subscribe from the webpage: http://www.asternic.org Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --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] Page about 70 users crash my Asterisk
On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote: I have here de backtrace result Using host libthread_db library /lib/libthread_db.so.1. Core was generated by `asterisk -g'. Program terminated with signal 11, Segmentation fault. #0 0xb7ece142 in ?? () As I see it was in the libthread library.. So can it confirm my theory that is a memory problem ? There's probably far more going on there than the initial backtrace you've got reveals. From doc/backtrace: This document is to provide information on how to obtain the backtraces required on the asterisk bug tracker, available at http://bugs.digium.com. The information is required by developers to help fix problem with bugs of any kind. Backtraces provide information about what was wrong when a program crashed; in our case, Asterisk. There are two kind of backtraces (aka 'bt'), which are useful: bt and bt full. First of all, when you start Asterisk, you MUST start it with option -g (this tells Asterisk to produce a core file if it crashes). If you start Asterisk with the safe_asterisk script, it automatically starts using the option -g. If you're not sure if Asterisk is running with the -g option, type the following command in your shell: debian:/tmp# ps aux | grep asterisk root 17832 0.0 1.2 2348 788 pts/1SAug12 0:00 /bin/sh /usr/sbin/safe_asterisk root 26686 0.0 2.8 15544 1744 pts/1SAug13 0:02 asterisk -vvvg -c [...] The interesting information is located in the last column. Second, your copy of Asterisk must have been built without optimization or the backtrace will be (nearly) unusable. This can be done by using 'make dont-optimize' intead of 'make install' to build and install the Asterisk binary and modules. After Asterisk crashes, a core file will be dumped in your /tmp/ directory. To make sure it's really there, you can just type the following command in your shell: debian:/tmp# ls -l /tmp/core.* -rw--- 1 root root 10592256 Aug 12 19:40 /tmp/core.26252 -rw--- 1 root root 9924608 Aug 12 20:12 /tmp/core.26340 -rw--- 1 root root 10862592 Aug 12 20:14 /tmp/core.26374 -rw--- 1 root root 9105408 Aug 12 20:19 /tmp/core.26426 -rw--- 1 root root 9441280 Aug 12 20:20 /tmp/core.26462 -rw--- 1 root root 8331264 Aug 13 00:32 /tmp/core.26647 debian:/tmp# Now that we've verified the core file has been written to disk, the final part is to extract 'bt' from the core file. Core files are pretty big, don't be scared, it's normal. *** NOTE: Don't attach core files on the bug tracker, we only need the bt and bt full. *** For extraction, we use a really nice tool, called gdb. To verify that you have gdb installed on your system: debian:/tmp# gdb -v GNU gdb 6.3-debian Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type show copying to see the conditions. There is absolutely no warranty for GDB. Type show warranty for details. This GDB was configured as i386-linux. debian:/tmp# Which is great, we can continue. If you don't have gdb installed, go install gdb. Now load the core file in gdb, as follows: debian:/tmp# gdb -se asterisk -c /tmp/core.26252 [...] (You would see a lot of output here.) [...] Reading symbols from /usr/lib/asterisk/modules/app_externalivr.so...done. Loaded symbols for /usr/lib/asterisk/modules/app_externalivr.so #0 0x29b45d7e in ?? () (gdb) Now at the gdb prompt, type: bt You would see output similar to: (gdb) bt #0 0x29b45d7e in ?? () #1 0x08180bf8 in ?? () #2 0xbcdffa58 in ?? () #3 0x08180bf8 in ?? () #4 0xbcdffa60 in ?? () #5 0x08180bf8 in ?? () #6 0x180bf894 in ?? () #7 0x0bf80008 in ?? () #8 0x180b0818 in ?? () #9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180 #10 0x00a0 in ?? () #11 0x00a0 in ?? () #12 0x in ?? () #13 0x407513c3 in confcall_careful_stream (conf=0x8180bf8, filename=0x8181de8 Zap/pseudo-1324221520) at app_meetme.c:262 #14 0x40751332 in streamconfthread (args=0x8180bf8) at app_meetme.c:1965 #15 0xbcdffbe0 in ?? () #16 0x40028e51 in pthread_start_thread () from /lib/libpthread.so.0 #17 0x401ec92a in clone () from /lib/libc.so.6 (gdb) The bt's output is the information that we need on the bug tracker. Now do a bt full as follows: (gdb) bt full #0 0x29b45d7e in ?? () No symbol table info available. #1 0x08180bf8 in ?? () No symbol table info available. #2 0xbcdffa58 in ?? () No symbol table info available. #3 0x08180bf8 in ?? () No symbol table info available. #4 0xbcdffa60 in ?? () No symbol table info available. #5 0x08180bf8 in ?? () No symbol table info available. #6 0x180bf894 in ?? () No symbol table info available. #7 0x0bf80008 in ?? () No symbol table info available. #8 0x180b0818 in ?? () No symbol table info available. #9 0x08068008 in ast_stopstream (tmp=0x40758d38) at file.c:180 No locals. #10
Re: [Asterisk-Users] Page about 70 users crash my Asterisk
On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote: Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls. We have program a page system with the page command and the auto answer funtion of polycom. We have detect via diaplan if the phone isn't in call we place the call. All this via Macro. But in the our that they are not many calls. So much user that can be page.. The Asterisk crash. We think it is a RAM Memory problem.. Do you have any idea for this ? It's nearly impossible to tell without a core dump or backtrace of the core file, but there have been a few key fixes to 1.2.X and /trunk recently in app_meetme that may solve a problem you're having here. Since app_page depends on app_meetme to function, you may want to upgrade to the latest version that's appropriate for you and then retest. -- Bird's The Word Technologies, Inc. http://www.btwtech.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] Page about 70 users crash my Asterisk
I have here de backtrace resultUsing host libthread_db library /lib/libthread_db.so.1.Core was generated by `asterisk -g'.Program terminated with signal 11, Segmentation fault. #0 0xb7ece142 in ?? ()As I see it was in the libthread library.. So can it confirmmy theory that is a memory problem ?On 3/23/06, BJ Weschke [EMAIL PROTECTED] wrote: On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote: Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls. We have program a page system with the page command and the auto answer funtion of polycom. We have detect via diaplan if the phone isn't in call we place the call. All this via Macro. But in the our that they are not many calls. So much user that can be page.. The Asterisk crash.We think it is a RAM Memory problem.. Do you have any idea for this ? It's nearly impossible to tell without a core dump or backtrace ofthe core file, but there have been a few key fixes to 1.2.X and /trunkrecently in app_meetme that may solve a problem you're having here. Since app_page depends on app_meetme to function, you may want toupgrade to the latest version that's appropriate for you and thenretest.--Bird's The Word Technologies, Inc. http://www.btwtech.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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] Page about 70 users crash my Asterisk
I have here de backtrace resultUsing host libthread_db library /lib/libthread_db.so.1.Core was generated by `asterisk -g'.Program terminated with signal 11, Segmentation fault. #0 0xb7ece142 in ?? ()As I see it was in the libthread library.. So can it confirmmy theory that is a memory problem ?On 3/23/06, BJ Weschke [EMAIL PROTECTED] wrote: On 3/23/06, Alvaro Parres [EMAIL PROTECTED] wrote: Hi list, i have and asterisk into a Pentium IV Server with 1GB of RAM about 75 Polycom Phones, one E1 for incoming calls. We have program a page system with the page command and the auto answer funtion of polycom. We have detect via diaplan if the phone isn't in call we place the call. All this via Macro. But in the our that they are not many calls. So much user that can be page.. The Asterisk crash.We think it is a RAM Memory problem.. Do you have any idea for this ? It's nearly impossible to tell without a core dump or backtrace ofthe core file, but there have been a few key fixes to 1.2.X and /trunkrecently in app_meetme that may solve a problem you're having here. Since app_page depends on app_meetme to function, you may want toupgrade to the latest version that's appropriate for you and thenretest.--Bird's The Word Technologies, Inc. http://www.btwtech.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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
