[asterisk-users] missing asterisk now rpm for centos5
Hi all, I’m not sure if this is the right list to send this to, but the “AsteriskNow” meta-package is missing from the centos/5/asterisk-1.8-certified/i386/RPMS directory. Is this package still available? I’ve got a VERY old machine that I am pressing into service; it won’t run CentOS 6. Many thanks Cassius Smith -- _ -- 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] Ast12 issue missing library file??
Hi ALL, still having trouble getting Ast 12 to run. I got it compiled and built but now when I try to run, I'm getting a missing library error that seems to be in error (see below). The .so file DOES exist with correct permissions. [root@Asterisk12 ~]# asterisk -rvvv asterisk: error while loading shared libraries: libasteriskssl.so.1: cannot open shared object file: No such file or directory BUT: [root@Asterisk12 ~]# find / -name libasteriskssl.so.1 /usr/lib/libasteriskssl.so.1 /usr/local/src/asterisk-12.0.0-beta1/main/libasteriskssl.so.1 [root@Asterisk12 ~]# ls -l /usr/lib/libasteriskssl.so* lrwxrwxrwx. 1 root root 19 Oct 21 16:08 /usr/lib/libasteriskssl.so - libasteriskssl.so.1 -rwxr-xr-x. 1 root root 625890 Oct 21 16:08 /usr/lib/libasteriskssl.so.1 [root@Asterisk12 ~]# Any ideas? Many thanks, Cassius -- _ -- 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] SOLVED: Asterisk12Beta- configure script/uuid missing??
On Fri, Oct 18, 2013 at 03:16:08PM -0400, Cassius Smith wrote: Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: … checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd and uuid-devel. No joy, still getting same error. Anyone else run into this? How did you get around it? libuuid-devel is what I think you need. As an aside, in the asterisk source there is an install_prereq script that can be used to install all the necessary packages for your platform: $ sudo contrib/scripts/install_prereq install Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer Thanks Shaun - the install_prereq script did the trick. Cassius -- _ -- 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] Asterisk12Beta- configure script/uuid missing??
Hello, I'm trying to build Asterisk12 on a Centos 6.4 VM. The configure script is erring out with: … checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) I have installed (using yum) uuid, uuidd and uuid-devel. No joy, still getting same error. Anyone else run into this? How did you get around it? cheers, Cassius Smith -- _ -- 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] Dialout from MeetMe to another conference (Asterisk 1.4)
On 10/10/11 10:40 AM, Josh Freeman cpe.jfree...@gmail.com wrote: Hello, I'm looking at a scenario in which, to make it work, I'd need to dial into a remote conference from within a local MeetMe room. That might include being able to dial a conference code after the call to the remote system was answered. *Ideally*, it would work such that I could dial a single extension from one of my local telephones which would both connect me to the local MeetMe room and also place an outbound call to the remote conference, log in, and connect that call to the local MeetMe room as well. It looks as though later versions of Asterisk have an Originate() application that would get me most of the way there, but I'm constrained to use an Asterisk 1.4 system which doesn't appear to have that application. Anyone have any ideas on how I might make something like this work? Regards, Josh Hey Josh, (curiosity) How come you can use only 1.4? Cassius -- _ -- 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] Linksys/Cisco 504G randomly restarts
Agree -- make sure you are at the latest firmware. ALSO: If you have provisioning enabled, and have a duplicate line in your xml files, that will cause a reboot. Cheers, Cassius Smith On 8/15/11 1:46 PM, C F shma...@gmail.com wrote: I have 3 Linksys/Cisco 504G phones they keep restarting at what seems to be random. Sometimes as short as 6 minutes. FW version is 7.4.3a I have searched and tried disabling FW check and all related settings. I also extended all the default 3600 resync checks to a lot longer. TIA CF -- _ -- 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] Receptionist Extension cannot be Pickup()'ed
Hello all,
I am struggling with an annoying problem. I have an installation with a
small number of Grandstream GXP2010 endpoints. Each endpoint has all the
extensions programmed into the phone for BLF - for instant pickup, transfer
or speed dial.
Except for the Receptionist phone, which is handled internally via the 0
extension. That extension drops into a [day-menu] context with an IVR after
the receptionist phone rings for 20 seconds.
The receptionist phone has a BLF field on all the other phones. But when
that phone rings, I think something is messing with some channel variable
that is preventing Pickup() from working.
ALL Other extensions can be picked up. ONLY the extension(s) I ring from the
day-menu cannot.
Here is a snip from my dialplan:
exten = s,1,NoOp()
same = n,Verbose(2,Processing incoming call from ${CALLERID(all)})
same = n(daycheck),GotoIfTime(08:30-16:59,mon-fri,*,*?open)
same = n,Set(MENU=night-menu)
same = n,Goto(night)
same = n(open),Set(MENU=day-menu)
same = n,Set(__PICKUPMARK=)
same = n,Dial(SIP/3100,20) ; 3100 is receptionists phone
; go to IVR if no answer
same = n,Goto(playmenu)
same = n(night),NoOp()
same = n(top),Wait(0.5)
same = n,GotoIf($[${COUNTER}=10]?wrong)
same = n(playmenu),Background(${MENU})
same = n(bypass),WaitExten(10)
; go straight to VM if they time out...
same = n,Goto(2,1)
same = n(wrong),Playback(something-terribly-wrong)
same = n,Playback(goodbye)
same = ,n,Hangup()
; within [day-menu] option 2 is Voicemail, option 1 is Directory.
=
Calls come in to the dialplan from the PSTN in the [from-pstn] context:
[from-pstn]
; catch analog phone call incoming, send it to main number
exten = s,1,Verbose(2,---Processing incoming call for ${EXTEN}
- in context from-pstn)
same = n,Answer() ; Wait for CallerID Spill
same = n,Wait(1.5) ; Wait for CallerID Spill
same = n,Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
same = n,NoOp()
same = n,Set(__PICKUPMARK=)
same = n,Goto(day-menu,s,1)
Calls are picked up via this context, included in [users]:
[BLF_group_pickup]
exten = _**31XX,1,Verbose(2,BLF Pickup Extension ${EXTEN})
same = n,Pickup(${EXTEN:2}@users${EXTEN:2}@default${EXTEN:2}@PICKUPMARK)
same = n,Hangup()
(I have also tried adding @day-menu to this, but it didn't work either).
Oh yes Asterisk v1.8.4.1, DAHDI 2.4.1.2 libpri 1.4.11.5
Thanks
Cassius
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Re: [asterisk-users] Receptionist Extension cannot be Pickup()'ed
top posting on purpose
I neglected to say all the extensions can be picked up remotely by the
other endpoints, EXCEPT the receptionist phone x3100. When calls go to that
station, they cannot be picked up. Sorry for the necessity to post twice.
/top posting on purpose
From: Cassius Smith cass...@cassius.org
Date: Fri, 05 Aug 2011 15:31:14 -0500
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Receptionist Extension cannot be Pickup()'ed
Hello all,
I am struggling with an annoying problem. I have an installation with a small
number of Grandstream GXP2010 endpoints. Each endpoint has all the extensions
programmed into the phone for BLF - for instant pickup, transfer or speed
dial.
Except for the Receptionist phone, which is handled internally via the 0
extension. That extension drops into a [day-menu] context with an IVR after
the receptionist phone rings for 20 seconds.
The receptionist phone has a BLF field on all the other phones. But when that
phone rings, I think something is messing with some channel variable that is
preventing Pickup() from working.
ALL Other extensions can be picked up. ONLY the extension(s) I ring from the
day-menu cannot.
Here is a snip from my dialplan:
exten = s,1,NoOp()
same = n,Verbose(2,Processing incoming call from ${CALLERID(all)})
same = n(daycheck),GotoIfTime(08:30-16:59,mon-fri,*,*?open)
same = n,Set(MENU=night-menu)
same = n,Goto(night)
same = n(open),Set(MENU=day-menu)
same = n,Set(__PICKUPMARK=)
same = n,Dial(SIP/3100,20) ; 3100 is receptionists phone
; go to IVR if no answer
same = n,Goto(playmenu)
same = n(night),NoOp()
same = n(top),Wait(0.5)
same = n,GotoIf($[${COUNTER}=10]?wrong)
same = n(playmenu),Background(${MENU})
same = n(bypass),WaitExten(10)
; go straight to VM if they time out...
same = n,Goto(2,1)
same = n(wrong),Playback(something-terribly-wrong)
same = n,Playback(goodbye)
same = ,n,Hangup()
; within [day-menu] option 2 is Voicemail, option 1 is Directory.
=
Calls come in to the dialplan from the PSTN in the [from-pstn] context:
[from-pstn]
; catch analog phone call incoming, send it to main number
exten = s,1,Verbose(2,---Processing incoming call for ${EXTEN} -
in context from-pstn)
same = n,Answer() ; Wait for CallerID Spill
same = n,Wait(1.5) ; Wait for CallerID Spill
same = n,Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
same = n,NoOp()
same = n,Set(__PICKUPMARK=)
same = n,Goto(day-menu,s,1)
Calls are picked up via this context, included in [users]:
[BLF_group_pickup]
exten = _**31XX,1,Verbose(2,BLF Pickup Extension ${EXTEN})
same = n,Pickup(${EXTEN:2}@users${EXTEN:2}@default${EXTEN:2}@PICKUPMARK)
same = n,Hangup()
(I have also tried adding @day-menu to this, but it didn't work either).
Oh yes Asterisk v1.8.4.1, DAHDI 2.4.1.2 libpri 1.4.11.5
Thanks
Cassius
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Re: [asterisk-users] References customers
What do you mean by customers? Are you looking for testimonials from satisfied users? -- On 7/10/11 11:53 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How can I find a references customers that used Asterisk as IP Telephony or Call Center or IVR? In which link they are mentioned? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] single keypress short-circuits to invalid extension handler
Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0throughdigits/0digit s/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0digits/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- _ -- 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] single keypress short-circuits to invalid extension handler
On 7/6/11 3:20 PM, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, July 06, 2011 4:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] single keypress short-circuits to invalid extension handler Hello all I'm running Asterisk 1.8.4.4 in a new installation. I'm seeing peculiar behaviour in a context where I dispatch to different MeetMe conference rooms. It seems the first digit is being given to Asterisk and it ALWAYS jumps to the i extension. I originally had single digits for the MeetMe rooms, I tried double digits to no avail. As soon as I press the 0 key it plays the invalid message. Here is my meet-me context from my dialplan. Any ideas? Other sections of my dialplan work fine in permitting multiple digit keypresses. I have used this same dialplan in many other installations, so I'm pretty flummoxed by this... Cassius Smith [meet-me] exten = s,1(top),NoOp() same = n,Answer() same = n,Wait(1.0) same = n,Background(enter-conf-call-numberdigits/0digits/0through digits/0digits/9) same = n,WaitExten(5) exten = 00,n,MeetMe(SouthAfrica0,dMs) exten = 01,n,MeetMe(Swaziland1,dMs) exten = 02,n,MeetMe(Botswana2,dMs) exten = 03,n,MeetMe(Zimbabwe3,dMs) exten = 04,n,MeetMe(Lesotho4,dMs) exten = 05,n,MeetMe(Mozambique5,dMs) exten = 06,n,MeetMe(Zimbabwe6,dMs) exten = 07,n,MeetMe(Namibia7,dMs) exten = 08,n,MeetMe(Angola8,dMs) exten = 09,n,MeetMe(Congo9,dMs) exten = t,1,Goto(s,top) exten = i,1,Playback(invalid) same = n,Goto(s,top) And here is the console output... -- Executing [4098@users:1] Goto(SIP/4099-0026, meet-me,s,1) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:2] Answer(SIP/4099-0026, ) in new stack -- Executing [s@meet-me:3] Wait(SIP/4099-0026, 1.0) in new stack -- Executing [s@meet-me:4] BackGround(SIP/4099-0026, enter-conf-call-numberdigits/0digits/0throughdigits/0dig its/9) in new stack -- SIP/4099-0026 Playing 'enter-conf-call-number.ulaw' (language 'en_ZA') -- Invalid extension '0' in context 'meet-me' on SIP/4099-0026 == CDR updated on SIP/4099-0026 -- Executing [i@meet-me:1] Playback(SIP/4099-0026, invalid) in new stack -- SIP/4099-0026 Playing 'invalid.slin' (language 'en_ZA') -- Executing [i@meet-me:2] Goto(SIP/4099-0026, s,top) in new stack -- Goto (meet-me,s,1) -- Executing [s@meet-me:1] NoOp(SIP/4099-0026, ) in new stack You don't have a priority 1 exten = 00,1,MeetMe(SouthAfrica0,dMs) exten = 01,1,MeetMe(Swaziland1,dMs) exten = 02,1,MeetMe(Botswana2,dMs) Etc. WaitExten can accept more than one digit. Thanks Eric - this was it. I knew WaitExten() would read more than 1 digit. I guess I'd been staring at it so long I couldn't see the error. I appreciate the extra eyes! Cassius -- _ -- 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] Cisco IP Phones 7942G (skinny): TFTP and required files
Hello, I do not use the skinny firmware. By the way, questions like this are best shared with the asterisk-users group mailing list, so that a large segment of the Asterisk community can benefit from the questions and answers. Cassius Smith -- On 6/16/11 4:59 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; I am sure that you have experience with Cisco IP Phones. I need to be sure if someone used Cisco 7942G in skinny firmware with Asterisk 1.8 and how it was (if fine or it has a problem). Are the below the only 3 needed files to be placed in the tftpboot directory: CTLSEPB8BEBF22AB62.tlv (which is empty file, just we place it with its name). SEPB8BEBF22AB62.cnf.xml XMLDefault.cnf.xml So, do I have to add any other file? One more thing: in the above mentioned files, do I have to determine the firmware that the Phone should take it and I have to place this firmware in the tftpboot directory? Note: I am using tftp-server (as my OS if fedora). Is there any permission need to be given for the files in the /var/lib/tftpboot/? Or no need as the phones are going to download them and not upload new files? Looking forward for a help PLZ. Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Page() bumps user out of a call
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})
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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] Asterisk 1.6.2.18, Cisco 79XX not registering
Did this fix make it into 1.8.4? Getting registration errors on Cisco 79XX in 1.8.4, going back to 1.8.3.3 everything works. I did open https://issues.asterisk.org/view.php?id=19264 and included a SIP trace. Sorry all, I did not follow up adequately. Definitely a problem with 1.6.2.18 and the issue # is 18951. Fixed in 1.8.3.3; Cisco 79xx registered fine. I don't know about 1.8.4 yet; haven't installed it for testing yet. Cassius This fix definitely not in 1.8.4; I also dropped back to 1.8.3.3 on a test box and Cisco 79XX's register correctly. Thanks for opening the issue; will check 1.8.5rc when it's available. Cassius On Fri, May 6, 2011 at 12:24 PM, Julian Lyndon-Smith aster...@dotr.com wrote: It was my problem ;) https://issues.asterisk.org/view.php?id=18951 fixed in svn On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote: On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com http://issues.digium.com and see if there is anything similar to your issue. I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- 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] lead time for RPM's?
Hi all Usually I build asterisk from source, but recently have been doing a couple of test installations with packages from the Digium repository. About how long does it take to get from new release announcement into the Digium RPM repository? Specifically 1.8.4 CentOS hasn't made it to the rpm repository yet. Cassius -- _ -- 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 1.6.2.18, Cisco 79XX not registering
Did this fix make it into 1.8.4? Getting registration errors on Cisco 79XX in 1.8.4, going back to 1.8.3.3 everything works. I did open https://issues.asterisk.org/view.php?id=19264 and included a SIP trace. Sorry all, I did not follow up adequately. Definitely a problem with 1.6.2.18 and the issue # is 18951. Fixed in 1.8.3.3; Cisco 79xx registered fine. I don't know about 1.8.4 yet; haven't installed it for testing yet. Cassius On Fri, May 6, 2011 at 12:24 PM, Julian Lyndon-Smith aster...@dotr.com wrote: It was my problem ;) https://issues.asterisk.org/view.php?id=18951 fixed in svn On 6 May 2011 16:45, Steve Davies davies...@gmail.com wrote: On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Friday, May 06, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious - has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com http://issues.digium.com and see if there is anything similar to your issue. I also remember this being mentioned - I believe it was fixed in the chan_sip Via: header handling code. The fix is in branches/1.6.2 already, so you should be able to grab the patch without too much trouble. Regards, Steve -- _ -- 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] Cisco 7940 phone and tftpd provisioning - for ever?
On 5/9/11 6:02 AM, Doug Lytle supp...@drdos.info wrote: Sebastian Arcus wrote: Cisco phones (at least the 7940) are supposed to be run with a tftp server available at all time That is my experience. But, if you're running tftp under Linux, then it's probably spawned by xinetd and won't be running unless the service is requested. Doug If you want the users to have access to ringtones and desktop images, they are dynamically loaded via tftp. So yes, you'll need to keep the tftp server running. HTH Cassius -- _ -- 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 1.6.2.18, Cisco 79XX not registering
Hi all, I have a production server running with about 90 Cisco 79[46]1's and SIP release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and upgraded last night after hours. (Seemed low risk to me!) Much to my surprise, not a single one of the Cisco 79XX phones would register. Since it's a production server, I rolled back to 1.6.2.9 and everything was fine. All my Linksys SPA phones and Polycom speaker phones registered just fine. I am now setting up test servers with both 1.6.2.18 and 1.8.3.3 to collect some debug. I am just curious has anyone else had SIP issues with these phones and updating Asterisk broke them? I will post results of my findings after I have time to collect them. Cassius Smitha -- _ -- 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] Play different voice-mail messages based on certain conditions
From: Warren Selby wcse...@selbytech.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Mon, 21 Mar 2011 20:37:52 -0500
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Play different voice-mail messages based on
certain conditions
On Mon, Mar 21, 2011 at 8:05 PM, Harel Cohen ha...@easycall.gi wrote:
Hello List,
I have few installations out there based on 1.6.1 or above.
I¹m trying to play different voice mail messages based on certain criteria¹s.
For example, I want during office hours to play (in short): ³we are not
available to take your call, please leave a message², during off-hours and
weekends I would play: ³we are closed, our opening hours xx:xx-yy:yy, please
leave a message or send a fax or send an email² and during holidays I would
play: ³we are closed due to holiday, please leave a message, fax, blab la²
etc.
What I have done for various clients in your situation is to create
conditional contexts based on either time of day and day of year criteria (see
GotoIfTime()[1]) and then use Playback() to play the correct voicemail
greeting, then call the Voicemail() app with just the s option, which skips
all vm-intro's and any pre-recorded messages.
Quick, off the top of my head example:
[default]
exten = _X.,1,Verbose(Incoming call - battlestations!)
exten = _X.,n,Answer()
exten = _X.,n,Dial(SIP/${EXTEN},30)
exten = _X.,n,Verbose(No one answered - going to voicemail)
exten = _X.,n,Goto(no-answer,s,1)
[no-answer]
; no one answered, play voicemail based on time of day / day of year
exten = s,1,Verbose(Checking time conditions to play proper voicemail)
exten = s,n,Verbose(First check holidays)
exten = s,n,GotoIfTime(*,*,25,dec?holiday,1) ; Christmas, add your own here
exten = s,n,Verbose(Not a holiday - so checking time of day)
exten = s,n,GotoIfTime(08:00-18:00,mon-fri,*,*?officehours,1)
exten = s,n,Verbose(Time condition check failed - playing after-hours
message)
exten = s,n,Goto(afterhours,1)
; holiday voicemail greeting
exten = holiday,1,Verbose(Playing holiday greeting)
exten = holiday,n,Playback(holiday-greeting)
exten = holiday,n,Voicemail(defaultmailbox@default,s)
exten = holiday,n,Hangup()
; officehours voicemail greeting
exten = officehours,1,Verbose(Playing officehours greeting)
exten = officehours,n,Playback(officehours-greeting)
exten = officehours,n,Voicemail(defaultmailbox@default,s)
exten = officehours,n,Hangup()
; afterhours voicemail greeting
exten = afterhours,1,Verbose(Playing afterhours greeting)
exten = afterhours,n,Playback(afterhours-greeting)
exten = afterhours,n,Voicemail(defaultmailbox@default,s)
exten = afterhours,n,Hangup()
[1]: http://www.voip-info.org/wiki/view/Asterisk+cmd+GotoIfTime
--
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
I used a slightly different approach below is a snippet of my [day-menu]
context that I did for one of my installs. I load up the site's holiday
schedule in the Asterisk data base (I do it once per year, and train someone
to do it if I cannot), then check for the current date/holiday key if it
is =1 then I play out the holiday greetings, otherwise I play out the day or
night greeting. The second argument to STRFTIME is the time zone be sure
to get that right for your installation.
[day-menu]
exten = s,1,Answer()
exten = s,n,Wait(1.5) ; Wait for callerID spill
exten = s,n,Set(DATE=${STRFTIME(${EPOCH},ChST,%C%y%m%d)})
exten = s,n,Verbose(2,--- Current date is ${DATE})
exten = s,n(reinit),Set(COUNTER=0)
exten = s,n,GotoIf($[${DB(custom/${DATE}/holiday)} = 1]?holiday)
exten = s,n,Goto(daycheck)
exten = s,n(holiday),Set(MENU=holiday-menu)
exten = s,n,Goto(playmenu)
exten = s,n(daycheck),GotoIfTime(08:00-16:59,mon-fri,*,*?open)
exten = s,n,Set(MENU=night-menu)
exten = s,n,Goto(night)
exten = s,n(open),Set(MENU=day-menu)
exten = s,n(night),NoOp()
exten = s,n(top),Wait(0.5)
exten = s,n,GotoIf($[${COUNTER}=10]?wrong)
exten = s,n(playmenu),Background(${MENU})
exten = s,n(bypass),WaitExten(10)
; go straight to VM if they time out...
exten = s,n,Goto(2,1)
exten = s,n(wrong),Playback(something-terribly-wrong)
exten = s,n,Playback(goodbye)
exten = s,n,Hangup()
Hopefully this is enough to get you started.
Cassius Smith
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Re: [asterisk-users] Need to buy the Digium card, to confirm
The X1 card should seat in the X4 or X8 slots. Check out: http://computer.howstuffworks.com/pci-express1.htm HTH Cassius Smith On 2/26/11 4:33 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; My server and its slots written in it the following so I need to know which card to order it (I need a card supporting 2 E1s): PCIE_G2_X4 PCIE_G2_X8 Actually I do not know what is meaning by G2. OK I tried to buy directly from the below link but I found it is mentioned that it is x1 and not x4 or x8 so how can I get x4 or x8? The link: http://store.digium.com/productview.php?product_code=TE220B Description for the product: Two (2) span digital T1/E1/J1/PRI PCI-Express x1 card So please advise what do to? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] no progress indication
On 2/18/11 5:18 PM, Paul Belanger pabelan...@digium.com wrote: On 11-02-18 03:59 PM, Cassius Smith wrote: I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). *CLI core show application Progress() CLI output: -- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036, SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack == Using SIP RTP CoS mark 5 [Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect [Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument -- Called RickEndpoint [Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument == Spawn extension (macro-StdExten, s, 6) exited non-zero on 'IAX2/barneveld-2036' in macro 'StdExten' == Spawn extension (no911, RickEndpoint, 1) exited non-zero on 'IAX2/barneveld-2036' -- Hungup 'IAX2/barneveld-2036' [Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument [Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission 367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. There is something going wrong here, netsock2 is not parsing the IP address correctly. Are you using realtime? It would be good to see a full debug[1] log of your call. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information Hi Paul, no, not using realtime. I collected the trace but it didn't seem to give much clue (at least to me). Here is an extract from the log (dialing extension 4511 this time). Let me know if you want the full debug log including IAX and SIP debugs. (trunk is IAX, endpoints are SIP). [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Macro' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [4511@no911:1] Macro(IAX2/barneveld-9539, StdExten,SIP/4511SIP/xlite-4511,20) in new stack [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'MACRO_EXTEN' is '4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Verbose' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:1] Verbose(IAX2/barneveld-9539, 2,Processing StdExten call for 4511) in new stack [Feb 20 00:23:23] VERBOSE[9962] app_verbose.c: == Processing StdExten call for 4511 [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Verbose [Feb 20 00:23:23] DEBUG[9962] pbx.c: Function result is 'Cassius Home 3703' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Verbose' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:2] Verbose(IAX2/barneveld-9539, 2,CallerID = Cassius Home 3703) in new stack [Feb 20 00:23:23] VERBOSE[9962] app_verbose.c: == CallerID = Cassius Home 3703 [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Verbose [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG1' is 'SIP/4511SIP/xlite-4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Set' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:3] Set(IAX2/barneveld-9539, Device=SIP/4511SIP/xlite-4511) in new stack [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Set [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'MACRO_EXTEN' is '4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Set' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:4] Set(IAX2/barneveld-9539, UserID=4511) in new stack [Feb 20 00:23:23] DEBUG[9962] app_macro.c: Executed application: Set [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG1' is 'SIP/4511SIP/xlite-4511' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Result of 'ARG2' is '20' [Feb 20 00:23:23] DEBUG[9962] pbx.c: Launching 'Dial' [Feb 20 00:23:23] VERBOSE[9962] pbx.c: -- Executing [s@macro-StdExten:5] Dial(IAX2/barneveld-9539, SIP/4511SIP/xlite-4511,20) in new stack [Feb 20
[asterisk-users] no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s@macro-StdExten:6] Dial(IAX2/barneveld-2036,
SIP/RickEndpointSIP/xlite-RickEndpoint,20) in new stack
== Using SIP RTP CoS mark 5
[Feb 18 20:43:04] WARNING[6160]: acl.c:698 ast_ouraddrfor: Cannot connect
[Feb 18 20:43:04] WARNING[6160]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
-- Called RickEndpoint
[Feb 18 20:43:04] WARNING[6160]: app_dial.c:2039 dial_exec_full: Unable to
create channel of type 'SIP' (cause 20 - Unknown)
[Feb 18 20:43:04] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:05] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:07] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
== Spawn extension (macro-StdExten, s, 6) exited non-zero on
'IAX2/barneveld-2036' in macro 'StdExten'
== Spawn extension (no911, RickEndpoint, 1) exited non-zero on
'IAX2/barneveld-2036'
-- Hungup 'IAX2/barneveld-2036'
[Feb 18 20:43:11] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:19] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:35] WARNING[3550]: chan_sip.c:3115 __sip_xmit: sip_xmit of
0x2aaad453c6c0 (len 798) to 0.0.17.159:5060 returned -1: Invalid argument
[Feb 18 20:43:36] WARNING[3550]: chan_sip.c:3386 retrans_pkt:
Retransmission timeout reached on transmission
367fd44f3a944b134765a4dc4c95b88d@127.0.0.1:5060 for seqno 102 (Critical
Request) -- See doc/sip-retransmit.txt.
Here is my StdExten macro:
[macro-StdExten]
exten = s,1,Verbose(2,Processing StdExten call for
${MACRO_EXTEN})
exten = s,n,Verbose(2,CallerID = ${CALLERID(all)})
exten = s,n,Set(Device=${ARG1})
exten = s,n,Set(UserID=${MACRO_EXTEN})
exten = s,n,Dial(${ARG1},${ARG2})
exten = s,n,Verbose(2,== Voicemail ${MACRO_EXTEN} -- unavail)
exten = s,n,Voicemail(${MACRO_EXTEN}@default,u)
exten = s,n,Hangup()
I was expecting the system to indicate that ringing was ?
I know I can check channel availability to bypass this behavior; just
curious why it's happening or whether it's expected.
Cassius
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Re: [asterisk-users] Question about EuroBRI final 2 digits
On 2/10/11 5:54 AM, Christian Gansberger christian.gansber...@accm.at wrote: Hello, Maybe try that: In your incoming isdn context: [isdn-incoming] exten = s,1,Set(TIMEOUT(digits)=3) exten = s,2,WaitExten(2) exten = s,3,Dial(SIP/operator...) exten = 10,1,Dial(SIP/10) exten = 20,1,Dial(SIP/20) So if a call comes in Asterisk waits, 2 seconds for further digits dialed and if so jumps to the right extension in the context. Overlapdial should be yes. yours christian gansberger www.accm.at Many thanks for this idea, Christian I have put this equivalent into the dialplan And when the Austria team gets to the office in the morning they will test it. (BTW changed TIMEOUT(digits) to TIMEOUT(digit)). Cassius On 3 February 2011 20:45, Cassius Smith cass...@cassius.org wrote: Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] Question about EuroBRI final 2 digits
Hello, I have an installation in Austria; ISDN service provided by Austria Telekom. The main number of the service is 6 digits. Incoming calls may contain 2 additional digits, which I then use to route the call to the correct extension. Telekom sends me all the digits. My problem is that when someone tries to dial an 8 digit number to an extension from an outside analog phone, AT sends the call before they finish dialing all 8 digits. Is there a way to prevent this, or to catch the additional 2 digits somewhere in the stream? The receptionist is unhappy because she gets all the 6-digit calls and must then transfer. Is this a p2p vs p2mp issue? Thanks in advance, Cassius Smith -- _ -- 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] TDM410 and DSL
Hi all, I have a system installation in Guam with two trunks. One has a DSL service riding on it with the usual filter. That channel however keeps throwing alarms. I bypassed the filter and it stopped throwing alarms, but of course the high frequencies annoy the users. I swapped the filters and the alarms came back. Any suggestions? Could I have a bad DSL modem? Cassius -- _ -- 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] Cisco IP Phones and AVAYA IP Phones: How to configure in Asterisk
CallFwd should be one of the soft keys on your Cisco phones. Are you re-flashing the Cisco phones with SIP? -Cassius On 1/2/11 3:50 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi All; How to configure the buttons in the Cisco IP Phones to be used for different functionalities like Call Forward, Call Pickup, ... etc? For example, if I need to assign one of the buttons existed at Cisco IP Phone to be used for CallFrw, how to do this in Asterisk? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA942 on speaker phone does not hang up?
That fixed it! THANK YOU. -Cassius From: Peder pe...@networkoblivion.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 24 Nov 2010 07:42:52 -0600 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA942 on speaker phone does not hang up? It is the phone itself: go to Regional tab and scroll down to Reorder Delay and make it 255. That tells it not to play re-order tone and just hangup. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, November 24, 2010 5:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SPA942 on speaker phone does not hang up? Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty simple for these extensions; here is what I am using now: [extension1234] mailbox=1...@default type=friend context=users host=dynamic secret=verysecret I have looked at the sample sip.conf and did not get any clues, also the SPA900 Admin Manual doesn't say anything about it. Thanks Cassius -- _ -- 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] SPA942 on speaker phone does not hang up?
Premature reply. It did fix the first issue. Now when I ring that phone I get busy here from the phone, and the call goes straight to voicemail per dialplan. Maybe another parameter in addition to Reorder Delay? From: Cassius Smith cass...@cassius.org Date: Thu, 25 Nov 2010 10:34:25 +0100 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA942 on speaker phone does not hang up? That fixed it! THANK YOU. -Cassius From: Peder pe...@networkoblivion.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Wed, 24 Nov 2010 07:42:52 -0600 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA942 on speaker phone does not hang up? It is the phone itself: go to Regional tab and scroll down to Reorder Delay and make it 255. That tells it not to play re-order tone and just hangup. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Wednesday, November 24, 2010 5:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SPA942 on speaker phone does not hang up? Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty simple for these extensions; here is what I am using now: [extension1234] mailbox=1...@default type=friend context=users host=dynamic secret=verysecret I have looked at the sample sip.conf and did not get any clues, also the SPA900 Admin Manual doesn't say anything about it. Thanks Cassius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA942 on speaker phone does not hang up?
Hello all, I am using Linksys SPA942 in my current installation activity. I see a peculiar behavior: A call is made and the SPA942 uses its speaker. When the far end of a call hangs up , the SPA942 stays off hook, and after a time plays a fast busy. The user then has to press the line presence button to hang up the phone. I think I must be missing some sip.conf parameter. My sip.conf is pretty simple for these extensions; here is what I am using now: [extension1234] mailbox=1...@default type=friend context=users host=dynamic secret=verysecret I have looked at the sample sip.conf and did not get any clues, also the SPA900 Admin Manual doesn't say anything about it. Thanks Cassius -- _ -- 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 and cisco 7970 - multiple lines
Post the germane portions of your xml. How does your phone register each line button? Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 22 Nov 2010 12:38:22 -0600 To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk and cisco 7970 - multiple lines I can¹t believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter -- _ -- 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] asterisk and cisco 7970 - multiple lines
I have done something similar; I am using SIP load 8.5.2. I use port 5060 on both line buttons. Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com Date: Mon, 22 Nov 2010 13:24:41 -0600 To: Cassius Smith cass...@cassius.org Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [asterisk-users] asterisk and cisco 7970 - multiple lines Below is my xml button 1 and button 2 portion. Any help will be appreciated. line button=1 featureID9/featureID featureLabelPete(260)/featureLabel proxyproxyip/proxy port5060/port name130/name displayNamePeter/displayName autoAnswer autoAnswerEnabled2/autoAnswerEnabled autoAnswerModeAuto Answer with Speakerphone/autoAnswerMode /autoAnswer callWaiting3/callWaiting authName130/authName authPasswordpass/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber850/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact7b452e87-4496-4762-e11f-b26751a1884b/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line line button=2 featureID9/featureID featureLabelIntercom/featureLabel proxyproxyip/proxy port5061/port name160/name displayNamePeter/displayName autoAnswer autoAnswerEnabled3/autoAnswerEnabled autoAnswerModeAuto Answer with Speakerphone/autoAnswerMode /autoAnswer callWaiting3/callWaiting authName160/authName authPasswordpass/authPassword sharedLinefalse/sharedLine messageWaitingLampPolicy3/messageWaitingLampPolicy messagesNumber850/messagesNumber ringSettingIdle4/ringSettingIdle ringSettingActive5/ringSettingActive contact7b452e87-4496-4762-e11f-b26751a1884b/contact forwardCallInfoDisplay callerNametrue/callerName callerNumberfalse/callerNumber redirectedNumberfalse/redirectedNumber dialedNumbertrue/dialedNumber /forwardCallInfoDisplay /line Thanks, Peter From: Cassius Smith [mailto:cass...@cassius.org] Sent: Monday, November 22, 2010 1:12 PM To: kowalla...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] asterisk and cisco 7970 - multiple lines Post the germane portions of your xml. How does your phone register each line button? Cassius From: Peter Kowalski kowalla...@gmail.com Organization: GreatValueMart Reply-To: kowalla...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 22 Nov 2010 12:38:22 -0600 To: asterisk-users@lists.digium.com Subject: [asterisk-users] asterisk and cisco 7970 - multiple lines I can¹t believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Door Contacts via Asterisk?
Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -- _ -- 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] Door Contacts via Asterisk?
Thanks to all for these replies. I appreciate the variety and this is a great example of the community supporting one another. I sent this in last night and awoke to a broad set of replies! Thanks all - I will post again once I decide on a solution. Cassius Smith On 11/15/10 9:09 PM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Mon, Nov 15, 2010 at 1:56 PM, jon pounder j...@inline.net wrote: On 11/15/2010 02:49 PM, Mark Scholten wrote: Anyone have a soft sip endpoint which can take touchtones over sip and run scripts ? that is a more general purpose integration solution to asterisk itself. I realize there are scripts for dialplans which can do this already but often the door is nowhere near the core asterisk server. Hello, We did something like that in the past (but for 1 company, but it shouldn¹t be really different). The easiest solution for us was to use a door opener that could work with almost any ³normall² phone connection and use a Linksys pap2t or something similar. With kind regards, Mark Scholten From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius Smith Sent: Monday, November 15, 2010 7:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Door Contacts via Asterisk? Hi all, I have had (what I consider) an odd request. The installation I'm working on now is an office on a multi-floor building. They 're looking for some kind of solution with the phone system to provide door control. We are a non-profit so of course I'm looking for something VERY inexpensive. I'm sure /someone/ has done something like this. I'd appreciate any ideas. Cassius Smith -- _ -- 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 Hey everyone, I just wanted to say good show to everyone who responded to this gentleman's request! -- _ -- 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] what interface for ISDN-10/20/30?
Hello all,I'm working with one of our offices (that is moving soon) and they're being offered ISDN-10/20/30 services from their TELCO. I'm wondering what kind of interface card I will need (I prefer using Digium's cards). Are the TE121/122/ or TE212/220 series cards compatible with this kind of service? Seems like the service would look like a PRI interface, but I'm not sure. The office is in Singapore.ThanksCassius Smith -- _ -- 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] IAX2 works one direction, but not the other...
BTW I apologize for the double send. -- _ -- 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] IAX2 works one direction, but not the other...
I'm having trouble getting an IAX2 connection between a couple of servers. Ican make calls from server B to server A, but when I call from Server A to serverB, I get "No authority found".On ServerA I am running Asterisk 1.6.2.9On ServerB I'm running 1.6.2.13Any hints for me? The registrations in both directions seem to work fine when I do an iax2 reload from the CLI.config file snips shown below.ThanksCassius Smith=On server B, I have the following:[general]register = serverB:longsecretpasswo...@servera_ip[serverA]type=friendhost=dynamicauth=md5secret=longsecretpassword1context=no911[serverB]type=friendhost=dynamicauth=md5secret=longsecretpassword2context=no911On server A, I have the following:[general]register = serverA:longsecretpasswo...@serverb_ip[serverB]type=friendhost=dynamicauth=md5secret=longsecretpassword2context=no911[cary]type=friendhost=dynamicauth=md5secret=longsecretpassword1context=no911 -- _ -- 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] advice re: Page() application
Hi all,I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down.To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up.I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith -- _ -- 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] advice re: Page() application
Thanks Mike - this does help. The setup will be a local server on the LAN, and hopefully have plenty of snort to handle the load (20-30 phones). I also am not quite ready to put out 1.8 for my users yet.Do you have a snippet of dialplan code you'd be willing to share to loop through a group and grab/build up a list of channels as you describe? That would be enlightening (and probably save me some time)!What I am hearing is - using a second line presence for the Page() function will work; auto-answer should work and I should only page the phones that are not in use.Cassius Original Message Subject: Re: [asterisk-users] advice re: Page() application From: "Mike" l...@net-wall.com Date: Thu, October 14, 2010 10:12 am To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.com Hi Cassius,Can`t help for SPA-942, but the Wiki had good info on the Polycoms. Use the Wiki and you`ll do good. Two warnings:1) It seems to me that the adhoc MeetMe room used by the page application slows things down quite a lot. If you page and have a phone nearby, you`ll hear yourself with quite a bit of delay. It`s very annoying if you`re paging and hearing the page at the same time. Apparently 1.8 supports multicast and will do this differently, but it’ll be a long while before I trust 1.8 to be stable enough for my needs.2) If you`re doing this over an Internet link (i.e. hosted PBX), keep in mind that because of the MeetMe (I imagine), even if the receiving phones aren’t creating audio, the bandwidth is still is used as if everyone was talking at the same time in a MeetMe room. No biggie if everything is on the LAN, but a bit of a problem if not and you have many phones.And here is a tip: auto-answer is good, but you`ll have to loop through every SIP registration on the phone before using Page() to see if they are being used before adding them to the Page. If not, the phone will not auto-answer (since you`re on a call already) but you`ll have a missed call everytime somebody pages you while you`re on the phone. Users hate that (with reason). You check if each and every phone is being used BEFORE adding them to your page. In other words, if 10 out of 15 phones are idle, Page() only those 10.Besides that, things work as advertised. MikeFrom: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cassius SmithSent: Wednesday, October 13, 2010 7:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] advice re: Page() applicationHi all,I'm planning a new Asterisk installation; the users want to duplicate the paging function they have with their current Panasonic hybrid system. They dial *3 and announce a held call on line 3, for example, and the announcements comes out of all the desktop phone speakers. I'm planning to implement this using the Page() application in addition to parking the call. The O'Reilly book doesn't talk much about Page(), just says that it dumps the channels into a dynamically created MeetMe room which is quickly torn down.To make this work with typical desktop speakerphones, is there anything I need to do in sip.conf? (I was thinking I might need to set autoanswer=yes, for example). I can use a second line presence on all the phones to support this if necessary; I'm using SPA-942s. I don't want all the phones to ring - just have the announcement audible at each phone without the user needing to pick up.I apologize for not being able to try this out myself - I'm out of the country with no access to sip phones right now. Any help/lessons learned using Page() would be most appreciated!Regards,Cassius Smith-- _ -- 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] 3rd party app store
Personally, I would like to see less commercial marketing on http://asterisk.org. I count 5 separate marketing ads on the download page alone. This is just my opinion. The level of commercialism on the Asterisk.org download page does not bother me at all. Seems eminently fair for Digium to advertise their free (!) entry points for Switchvox and FFA. Asterisk training support - I have no problem with those either. The support and training are pay-for products, but are a big help to the community also. My $0.02. Cassius Smith -- _ -- 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] Help me Out!!!!
Clearly, if Word cannot explain the anguish in his heart, Mr. Fugina should be using OpenOffice! Cheers. -- _ -- 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] High volume BLF - Suggestions?
Steve I have 64 channels being monitored with an SPA962 with two SPA932 sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very happy with this. Latest firmware is a must. HTH Cassius Smith -- _ -- 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] Dahdi install gone wrong
* -Original Message- * From: Todd Reese trees...@gmail.com * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com * To: asterisk-users@lists.digium.com * Subject: [asterisk-users] Dahdi install gone wrong * Date: Mon, 23 Aug 2010 10:26:58 -0400 * * Hi All, * * * I've got a project installing a Digium TDM800P card with 8 FXO's in an * Asterisk box. * * * The computer is running Slackware 13.1 and I've installed the current * Dahdi and Asterisk 1.6.2.11. * * * I've installed several boxes that are pure VOIP but, I haven't installed * a Dahdi interface and I'm stumped. I've got it to the point of Dahdi * seeing the card and Asterisk recognizing dahdi but, I can't see any * channels for the calls to come in on. * * I've had to borrow files from an old config of Trixbox (the machine was * underpowered) to get to the point where I am in my setup. * * I would like to inquire some help from the group to get me up and * receiving calls on the card. * * * Regards, * * Todd Reese * * Include: * * * chan_dahdi.conf== * * * ; Configuration file * * [trunkgroups] * * [channels] * * language=en * context=from-zaptel * signalling=fxs_ks * rxwink=300 ; Atlas seems to use long (250ms) winks * ; * ; Whether or not to do distinctive ring detection on FXO lines * ; * ;usedistinctiveringdetection=yes * * usecallerid=yes * hidecallerid=no * callwaiting=yes * usecallingpres=yes * callwaitingcallerid=yes * threewaycalling=yes * transfer=yes * cancallforward=yes * callreturn=yes * echocancel=yes * echocancelwhenbridged=no * ;echotraining=800 * rxgain=0.0 * txgain=0.0 * group=0 * callgroup=1 * pickupgroup=1 * immediate=no * * ;faxdetect=both * faxdetect=incoming * ;faxdetect=outgoing * ;faxdetect=no * * ;Include setup-pstn configs * #include dahdi-channels.conf * * group=1 * * ;Include PBXconfig configs * #include chan_dahdi_additional.conf * * * * dahdi-channels.conf= * * ; Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 18 20:25:02 2010 * ; If you edit this file and execute /usr/sbin/dahdi_genconf again, * ; your manual changes will be LOST. * ; Dahdi Channels Configurations (chan_dahdi.conf) * ; * ; This is not intended to be a complete chan_dahdi.conf. Rather, it is * intended * ; to be #include-d by /etc/chan_dahdi.conf that will include the global * settings * ; * * ; Span 1: WCTDM/0 Wildcard TDM800P Board 1 (MASTER) * ;;; line=1 WCTDM/0/0 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 1 * callerid= * group= * context=default * * ;;; line=2 WCTDM/0/1 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 2 * callerid= * group= * context=default * * ;;; line=3 WCTDM/0/2 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 3 * callerid= * group= * context=default * * ;;; line=4 WCTDM/0/3 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 4 * callerid= * group= * context=default * * ;;; line=5 WCTDM/0/4 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 5 * callerid= * group= * context=default * * ;;; line=6 WCTDM/0/5 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 6 * callerid= * group= * context=default * * ;;; line=7 WCTDM/0/6 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 7 * callerid= * group= * context=default * * ;;; line=8 WCTDM/0/7 FXSKS (SWEC: MG2) * signalling=fxs_ks * callerid=asreceived * group=0 * context=from-pstn * channel = 8 * callerid=
Re: [asterisk-users] Caller ID issue
Sorry for the delay - I lost this message in the middle of a digest. I tried Answer(2000) and was getting an annoying warning: [Aug 15 17:20:11] WARNING[15516]: channel.c:1044 __ast_queue_frame: Exceptionally long voice queue length queuing to DAHDI/1-1 So I changed it back to Wait(2). I'll try shorter wait intervals and see what happens. Cassius Subject: Re: [asterisk-users] Caller ID issue To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: aanlkti=s6fboeqysvpvw25tmevkdpnjbvjmsvniwu...@mail.gmail.com Content-Type: text/plain; charset=ISO-8859-1 In most cases wait(.5) will do. I would not recommend using answer(2000) as that answers the channel, which means you start getting billed. On 8/2/10, Peder pe...@networkoblivion.com wrote: I am using T1's and didn't think the spill would take that long. PRI no, EM yes. Some PRI take that long too because the telco sends the name in a followup message, not in the initial call setup. -- -- _ -- 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] BLF/Call Pickup using SPA942, SPA962, SPA932
After chasing this some more, I decided to do the following: 1. Change the pickup code on the phone to *8# 2. Add an extension as follows: exten = _*8XXX,1,Pickup($EXTEN:2}) This worked. When I first tried it, I included a context but that didn't work for me (could be my dialplan context includes). Cassius -Original Message- From: Cassius Smith cass...@cassius.org Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932 Date: Sat, 14 Aug 2010 23:02:06 -0500 Yes, all set to same pickup group. Here is sip.conf setup (all ext's are similarly configured): [600] type=friend mailbox=...@default context=users pickupgroup=1 host=dynamic secret=*** -Original Message- From: Ron nha...@gmail.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932 Date: Sun, 15 Aug 2010 07:29:11 +0800 hi, just taking a wild guess here, are the extensions set to be in the same pickupgroup? regards ron On 8/15/10 7:01 AM, Cassius Smith wrote: Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf has *8 as the pickupexten in features.conf. On the SPA's the extended function is: fnc=blf+sd+cp;sub=...@$proxy;ext=...@$proxy the SPA932 Call Pickup Code: field is set to *8. I ring the extension; the lamp flashes on the shared line on the SPA, just like it should. When I press the flashing lamp, the CLI gives me: Notice [1328] Nothing to pick up for baf8bc-e23bc...@192.168.1.39 note: (this is the ip address of the SPA-942 in this case) then Got SIP response 603 Decline back from 192.168.1.47 note: (this is the ringing extension, in this case a Polycom 330). I have tried different pickup codes, and some web pages say to add a # at the end of the call pickup code. When I do that, the CLI says Notice [1328] Call from '602' to extension '**600' rejected because extension not found So - how to resolve this? Do I need dialplan code to handle this? I get the clue from nothing to pickup for blah blah that I'm close but may be missing something simple. Thanks all Cassius -- _ -- 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] BLF/Call Pickup using SPA942, SPA962, SPA932
Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf has *8 as the pickupexten in features.conf. On the SPA's the extended function is: fnc=blf+sd+cp;sub=...@$proxy;ext=...@$proxy the SPA932 Call Pickup Code: field is set to *8. I ring the extension; the lamp flashes on the shared line on the SPA, just like it should. When I press the flashing lamp, the CLI gives me: Notice [1328] Nothing to pick up for baf8bc-e23bc...@192.168.1.39 note: (this is the ip address of the SPA-942 in this case) then Got SIP response 603 Decline back from 192.168.1.47 note: (this is the ringing extension, in this case a Polycom 330). I have tried different pickup codes, and some web pages say to add a # at the end of the call pickup code. When I do that, the CLI says Notice [1328] Call from '602' to extension '**600' rejected because extension not found So - how to resolve this? Do I need dialplan code to handle this? I get the clue from nothing to pickup for blah blah that I'm close but may be missing something simple. Thanks all Cassius -- _ -- 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] BLF/Call Pickup using SPA942, SPA962, SPA932
Yes, all set to same pickup group. Here is sip.conf setup (all ext's are similarly configured): [600] type=friend mailbox=...@default context=users pickupgroup=1 host=dynamic secret=*** -Original Message- From: Ron nha...@gmail.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932 Date: Sun, 15 Aug 2010 07:29:11 +0800 hi, just taking a wild guess here, are the extensions set to be in the same pickupgroup? regards ron On 8/15/10 7:01 AM, Cassius Smith wrote: Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf has *8 as the pickupexten in features.conf. On the SPA's the extended function is: fnc=blf+sd+cp;sub=...@$proxy;ext=...@$proxy the SPA932 Call Pickup Code: field is set to *8. I ring the extension; the lamp flashes on the shared line on the SPA, just like it should. When I press the flashing lamp, the CLI gives me: Notice [1328] Nothing to pick up for baf8bc-e23bc...@192.168.1.39 note: (this is the ip address of the SPA-942 in this case) then Got SIP response 603 Decline back from 192.168.1.47 note: (this is the ringing extension, in this case a Polycom 330). I have tried different pickup codes, and some web pages say to add a # at the end of the call pickup code. When I do that, the CLI says Notice [1328] Call from '602' to extension '**600' rejected because extension not found So - how to resolve this? Do I need dialplan code to handle this? I get the clue from nothing to pickup for blah blah that I'm close but may be missing something simple. Thanks all Cassius -- _ -- 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] Caller ID issue
Hi list,
I'm having a problem with CallerID names not showing up when calls come
in. I have dialplan code to store the callerid(name) away and it is
blank (null). However, the voicemail variable ${VM_CALLERID} has the
name field populated. For example, here is some of the dialplan code:
2. Set(CALLER_ID_INFO_ALL=${CALLERID(all)})
3. Set(CALLER_ID_INFO_NAME=${CALLERID(name)})
4. Set(CALLER_ID_INFO_NUM=${CALLERID(num)})
5. Set(CALLER_ID_INFO_ANI=${CALLERID(ANI)})
6. Set(CALLER_ID_INFO_DNID=${CALLERID(DNID)})
Which yields this at the CLI:
-- Executing [3...@from_outside:2] Set(DAHDI/1-1,
CALLER_ID_INFO_ALL= 2565551212) in new stack
-- Executing [3...@from_outside:3] Set(DAHDI/1-1,
CALLER_ID_INFO_NAME=) in new stack
-- Executing [3...@from_outside:4] Set(DAHDI/1-1,
CALLER_ID_INFO_NUM=2565551212) in new stack
-- Executing [3...@from_outside:5] Set(DAHDI/1-1,
CALLER_ID_INFO_ANI=2565551212) in new stack
Note the first line should have the name field with the number, but does
not.
HOWEVER the voicemail notification contains:
Just wanted to let you know you were just left a 0:04 long message
(number 1) in mailbox 3703 from SMITH CASSIUS 2565551212
So - I know the NAME field is getting into the system, but it's not
showing up on the phones (and with telemarketers, that annoys my
users).
I'm using Asterisk 1.6.2.9, DAHDI 2.3.0
I have added callerid=asreceived to chan_dahdi.conf for my inbound
trunks, and shrinkcallerid=no to my sip.conf. (without effect)
Any ideas?
THANKS
Cassius
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Re: [asterisk-users] Caller ID issue
Thanks Warren. That fixed it. I am using T1's and didn't think the spill would take that long. Ciao, Cassius Add a Wait(2) before your first Set statement. Sometimes callerid takes a few seconds to arrive over the line, depending on your technology. -- _ -- 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] Peculiar Polycom IP6000 behavior
Here's a strange thing. I'm deploying Asterisk 1.6.2.9 with a pile of Cisco 79xx phones. For conference rooms we're using Polycom IP6000's. We bought two of them brand new. When I configure one phone with a username(SPIDR-3758)/password , it works fine. The other phone won't register with it's user(SPIDR-3749)/pass pair. When I try to use the first phone with the second user/pass pair, it won't work with that pair either. So, you'd think something was wrong with my sip.conf. I deleted the second entry and re-did it with new text. Still no joy. [SPIDR-3758](caryspider) mailbox=3...@default The above entry works, but: [SPIDR-3749](caryspider) mailbox=3...@default This one doesn't. [caryspider] looks like this: [caryspider](!) type=friend context=users host=dynamic secret=xx Any ideas? I'm stumped. Cassius Smith -- _ -- 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] Does SIP limit to 3-way conference?
Hello all, I'm in final testing stages and preparing training for a new Asterisk rollout. I'm replacing a Cisco Call Manager system, and re-flashing the 79x1 phones with SIP 8.5.2. With the SIP load and while in a call, I use the Confrn softkey to invite other participants. I can add one other participant endpoint into the conference, but no more. I know I can (and will) use MeetMe to do large conferences. My question is - am I forced to do so by SIP? Or am I missing something? Thanks! Cassius Smith -- _ -- 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] problem with voicemail contexts
Hello all, I am putting together an installation for our organization. My dialplan has most users in context [inside], and a separate [users] context includes the inside context. My voicemail config file has these users in a [users] context. I did this so I could get the name directory to work and vector calls to the right extensions. Now, however, I don't get message waiting lamp to show up on the phones and when the recipient of a voicemail tries to retrieve the message Alyson says you have no messages. This is true. The message doesn't get moved into the INBOX directory for the mailbox. I am flummoxed. Any ideas welcome! Cassius Smith -- _ -- 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] problem with voicemail contexts
OK, feeling very stupid right now. The test mailbox had delete=yes option set. All cleared up; sorry for cluttering up the list. Cassius snip Now, however, I don't get message waiting lamp to show up on the phones and when the recipient of a voicemail tries to retrieve the message Alyson says you have no messages. This is true. The message doesn't get moved into the INBOX directory for the mailbox. snip -- _ -- 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
