[asterisk-users] Grandstream Busy Light Fields
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can
see the green (free) status and red (busy) status.
What I cannot do is to accept the call when someone rings a remote
extension. The BLF button starts to blink in red telling me that the call is
ringing on remote extenson, but if I press it, my phone starts ringing that
extension instead of accepting the call. I can only acept the call from
other sides if I enter *8 (send) button.
Here I am sending an example from my sip.conf file (I have sip accounts from
60-90) :
[62]
; telefon
Callgroup=3
pickupgroup=1-20
subscribecontext=BLF
context=buster
type=friend
username=62
secret=secret
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
canreinvite=no
callerid=62
insecure=port
host=dynamic
dtmfmode=rfc2833
rtptimeout=10
nat=no
call-limit=20
[63]
; telefon
Callgroup=4
pickupgroup=1-20
subscribecontext=BLF
context=buster
type=friend
username=63
secret=secret
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
canreinvite=no
callerid=63
insecure=port
host=dynamic
dtmfmode=rfc2833
rtptimeout=10
nat=no
call-limit=20
[64]
; telefon
Callgroup=5
pickupgroup=1-20
subscribecontext=BLF
context=buster
type=friend
username=64
secret=secret
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
canreinvite=no
callerid=64
insecure=port
host=dynamic
dtmfmode=rfc2833
rtptimeout=10
nat=no
call-limit=20
Here the output from extensions.conf:
[buster]
; the usual dialplan+extensions are here, then comes:
; Pickup BLF
exten = _**6,1,Pickup(${EXTEN:1})
exten = _**6,2,Hangup
exten = _**11,1,Pickup(${EXTEN:2})
exten = _**11,2,Hangup
exten = _**6X,1,Pickup(${EXTEN:2})
exten = _**6X,2,Hangup
exten = _**7X,1,Pickup(${EXTEN:2})
exten = _**7X,2,Hangup
exten = _**8X,1,Pickup(${EXTEN:2})
exten = _**8X,2,Hangup
[BLF]
exten = 60,hint,SIP/60 ; Jozi
exten = 61,hint,SIP/61 ; Fax
exten = 62,hint,SIP/62 ; Tomaz
exten = 63,hint,SIP/63 ; Luka
exten = 64,hint,SIP/64 ; Petra
exten = 65,hint,SIP/65 ; Primoz
exten = 66,hint,SIP/66 ; Tibor
exten = 67,hint,SIP/67 ; Gregor
exten = 68,hint,SIP/68 ; Bostjan
exten = 69,hint,SIP/69 ; Oskar
exten = 70,hint,SIP/70 ; Jan
exten = 71,hint,SIP/71 ; Sinisa
exten = 73,hint,SIP/73 ; Tomaz Doma
exten = 78,hint,SIP/78 ; Tomaz Ratece
exten = 80,hint,SIP/80 ; Bostjan doma
exten = 82,hint,SIP/82 ; Tomaz
exten = 92,hint,SIP/92 ; Tomaz
exten = 95,hint,SIP/95 ; Test
This is it.
Kind regards,
Jan Prunk
--
Jan Prunk janprunk AT SPAMFREE gmail DOT com
Website: http://www.prunk.si PGP key: 00E80E86
Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86
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Re: [asterisk-users] Website callback
On Wed, 18 Jun 2008, Mark Hamilton wrote: Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as to exactly what time Asterisk will need to call. Then, Asterisk calls it (by way of call files? Either putting the call file in at the time it needs to be called, or I don't know what else) and then if the call is has a human on it, plays a message saying We're now transferring you to an agent. Please wait. And transfer that call to a queue. How can I do this? Is there something prebuilt like this? Reading the replies so-far... Cron jobs, databases, shell scripts... Ye Gods... Try reading the manual (or at least the wiki) http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Scroll down to the bit headed: How to schedule a Call in the Future Assuming you already have some PHP to write the call-file and move it into place, inserting a touch call after writing the file and moving it into place ought to be trivial... http://uk.php.net/manual/en/function.touch.php Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
Jan Prunk wrote: Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it, my phone starts ringing that extension instead of accepting the call. I can only acept the call from other sides if I enter *8 (send) button. Here I am sending an example from my sip.conf file (I have sip accounts from 60-90) : I may have it wrong, but I've found that PickUp can be a complete pain in the arse. In the same arrangement, I have an agi script called for (what in yours would be _**6X,1, which picks up a group of extensions (well all 4 lines on the [EMAIL PROTECTED] (internal queue context) and the extension number itself. This seems to work for incoming calls, but not internal calls. (Although I haven't really looked into this, I do know that PickUp doesn't work ifn certain circumstances, such as the extension being picked up using a Macro to which handles the dialling for instance). Now hopefully someone will come along and explain all the bits I got wrong. :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
On Thu, 19 Jun 2008, Jan Prunk wrote:
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can
see the green (free) status and red (busy) status.
Firstly, make sure the GS phones are of a relatively new hardware revision
and have the latest firmware loaded...
Program-- 1.1.6.16Bootloader-- 1.1.6.5
is a current good version for GXP phones.
Here the output from extensions.conf:
[buster]
; the usual dialplan+extensions are here, then comes:
; Pickup BLF
exten = _**6,1,Pickup(${EXTEN:1})
exten = _**6,2,Hangup
This is wrong - you need :2 to skip over the 2 stars.
exten = _**11,1,Pickup(${EXTEN:2})
exten = _**11,2,Hangup
exten = _**6X,1,Pickup(${EXTEN:2})
exten = _**6X,2,Hangup
exten = _**7X,1,Pickup(${EXTEN:2})
exten = _**7X,2,Hangup
exten = _**8X,1,Pickup(${EXTEN:2})
exten = _**8X,2,Hangup
You might want to try:
exten = _**.,1,Pickup(${EXTEN:2})
exten = _**.,n,Hangup()
Which is what I use and it works just fine.
And one other thing to remember if it's relevant for you, is that
different phones send different things - eg. Snoms send *8 then the
extension number, so putting:
exten = _**.,1,Pickup(${EXTEN:2})
exten = _**.,n,Hangup()
exten = _*8.,1,Pickup(${EXTEN:2})
exten = _*8.,n,Hangup()
Somewhere covers both cases.
Gordon
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Re: [asterisk-users] Mapping multimedia keys: pressed key not recognized
On Wed, Jun 18, 2008 at 08:21:06PM -0400, OCG Technical Support wrote: I've tried a few approaches to making the multimedia keys on my kbd play nice with myth, but all have lead to dead ends. One such dead end is to post this question to the Asteris Users mailing list, I guess :-( Wrong list? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
On Thu, Jun 19, 2008 at 09:22:04AM +0100, Gordon Henderson wrote: Reading the replies so-far... Cron jobs, databases, shell scripts... Ye Gods... Try reading the manual (or at least the wiki) http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Scroll down to the bit headed: How to schedule a Call in the Future What's the performance impact of having e.g. 30 such files at any given moment in the spool directory? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error: conflicting types for ‘bool’
On Wed, Jun 18, 2008 at 02:02:28PM -0700, Robert McNaught wrote: Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos 5, and getting the following error trail on make. Googling the issue has found one user who tried: seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Roman Yet Another Backporting mess! Bah! Yes, it should be safe to do so. The only thing non-standard with the machine is a RAID controller which was installed from a floppy in the first part of installing linux linux dd from the anaconda prompt. I have since done a yum -y update kernel kernel-devel and rebooted the machine. The running kernel is the same as the sources Linux localhost.localdomain 2.6.18-92.1.1.el5 #1 SMP Thu May 22 09:01:47 EDT 2008 x86_64 x86_64 x86_64 GNU/Linux Does anyone have any troubleshooting advice on this? Thanks in advance Robert .o wct4xxp/ wctc4xxp/ xpp/ wctdm24xxp/ wcte12xp/ modules make[2]: Entering directory `/usr/src/kernels/2.6.18-92.1.1.el5-x86_64' CC [M] /usr/src/zaptel-1.4.11/kernel/wctdm24xxp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wctdm24xxp/wctdm24xxp.o CC [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/../voicebus.o LD [M] /usr/src/zaptel-1.4.11/kernel/wcte12xp/wcte12xp.o CC [M] /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /usr/src/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /usr/src/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for ‘bool’ include/linux/types.h:36: error: previous declaration of ‘bool’ was here make[4]: *** [/usr/src/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 make[3]: *** [/usr/src/zaptel-1.4.11/kernel/xpp] Error 2 make[2]: *** [_module_/usr/src/zaptel-1.4.11/kernel] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-92.1.1.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/zaptel-1.4.11' make: *** [all] Error 2 ___ -- 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 -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding ;password=foo;method=bar to SIP uri
19 jun 2008 kl. 00.34 skrev Tom Browning:
To send calls into a custom SER implementation, I need to be able to
add some items to the URI that Asterisk will then send as part of
the INVITE
Asterisk dial SIP/[EMAIL PROTECTED]
needs to become
Asterisk dial SIP/[EMAIL PROTECTED];password=foo;method=bar
This is not a registration password. It is a passsword associated
with the destination xyz at location abc.com
Asterisk 1.4.18.1 seems to glue the data as part of the hostname and
fail to lookup abc.com
Is there a way to manipulate the URI that will be sent in the INVITE
to accomplish this?
From doc/channelvariables.txt
${SIP_URI_OPTIONS} * additional options to add to the URI for an
outgoing call
/Olle :-)
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
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Re: [asterisk-users] [FreeBSD 6.3] Zaptel stops responding
On Wed, 18 Jun 2008 12:47:04 -0500, Tilghman Lesher [EMAIL PROTECTED] wrote: Please call the reseller from which you bought the card or the manufacturer for support. Will do, although it could be a problem in the Zaptel code, which is not written by the mfg. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
On Thu, 19 Jun 2008, Tzafrir Cohen wrote: On Thu, Jun 19, 2008 at 09:22:04AM +0100, Gordon Henderson wrote: Reading the replies so-far... Cron jobs, databases, shell scripts... Ye Gods... Try reading the manual (or at least the wiki) http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Scroll down to the bit headed: How to schedule a Call in the Future What's the performance impact of having e.g. 30 such files at any given moment in the spool directory? A very quick scan through pbx_spool.c would suggest it's reasonably efficient as it only does an opendir/readdir at most once a second, and even then, only if the directory itself has been updated since the last look - there also seems to be more code to do a scan if it's not been updated, but when a file is old enough... I think Maybe the author could comment deeper :) So the inneficiencies will come from the underlying operating system when it's doing the actual readdir operation - which for 30-1000 files isn't going to be that high - less-so if if it's using ext3 + btree/dir_index mode. (or some other filesystem with efficient directory searches) Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
Try adding you context in that the phone is subscribed to. I had some issue
with this because if you do not specify the context it defaults to “default”
and has trouble finding the phone correctly. If you watch your debug very
closely I you should see it try to pick the phone up in the wrong context. In
your case the phone registers in the context “buster” so you would want your
BLF pickup to look like this:
exten = _**6X,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _**6X,2,Hangup
Hope this helps you out.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Prunk
Sent: Thursday, June 19, 2008 2:23 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Grandstream Busy Light Fields
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can see
the green (free) status and red (busy) status.
What I cannot do is to accept the call when someone rings a remote extension.
The BLF button starts to blink in red telling me that the call is ringing on
remote extenson, but if I press it, my phone starts ringing that extension
instead of accepting the call. I can only acept the call from other sides if I
enter *8 (send) button.
Here I am sending an example from my sip.conf file (I have sip accounts from
60-90) :
[62]
; telefon
Callgroup=3
pickupgroup=1-20
subscribecontext=BLF
context=buster
type=friend
username=62
secret=secret
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
canreinvite=no
callerid=62
insecure=port
host=dynamic
dtmfmode=rfc2833
rtptimeout=10
nat=no
call-limit=20
[63]
; telefon
Callgroup=4
pickupgroup=1-20
subscribecontext=BLF
context=buster
type=friend
username=63
secret=secret
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
canreinvite=no
callerid=63
insecure=port
host=dynamic
dtmfmode=rfc2833
rtptimeout=10
nat=no
call-limit=20
[64]
; telefon
Callgroup=5
pickupgroup=1-20
subscribecontext=BLF
context=buster
type=friend
username=64
secret=secret
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
canreinvite=no
callerid=64
insecure=port
host=dynamic
dtmfmode=rfc2833
rtptimeout=10
nat=no
call-limit=20
Here the output from extensions.conf:
[buster]
; the usual dialplan+extensions are here, then comes:
; Pickup BLF
exten = _**6,1,Pickup(${EXTEN:1})
exten = _**6,2,Hangup
exten = _**11,1,Pickup(${EXTEN:2})
exten = _**11,2,Hangup
exten = _**6X,1,Pickup(${EXTEN:2})
exten = _**6X,2,Hangup
exten = _**7X,1,Pickup(${EXTEN:2})
exten = _**7X,2,Hangup
exten = _**8X,1,Pickup(${EXTEN:2})
exten = _**8X,2,Hangup
[BLF]
exten = 60,hint,SIP/60 ; Jozi
exten = 61,hint,SIP/61 ; Fax
exten = 62,hint,SIP/62 ; Tomaz
exten = 63,hint,SIP/63 ; Luka
exten = 64,hint,SIP/64 ; Petra
exten = 65,hint,SIP/65 ; Primoz
exten = 66,hint,SIP/66 ; Tibor
exten = 67,hint,SIP/67 ; Gregor
exten = 68,hint,SIP/68 ; Bostjan
exten = 69,hint,SIP/69 ; Oskar
exten = 70,hint,SIP/70 ; Jan
exten = 71,hint,SIP/71 ; Sinisa
exten = 73,hint,SIP/73 ; Tomaz Doma
exten = 78,hint,SIP/78 ; Tomaz Ratece
exten = 80,hint,SIP/80 ; Bostjan doma
exten = 82,hint,SIP/82 ; Tomaz
exten = 92,hint,SIP/92 ; Tomaz
exten = 95,hint,SIP/95 ; Test
This is it.
Kind regards,
Jan Prunk
--
Jan Prunk janprunk AT SPAMFREE gmail DOT com
Website: http://www.prunk.si PGP key: 00E80E86
Fingerprint: 77C5156E29A4EB6C1C4A5EBA414A29F500E80E86
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Re: [asterisk-users] error: conflicting types for ‘bool’
On Wed, Jun 18, 2008 at 02:02:28PM -0700, Robert McNaught wrote: Trying to install zaptel-1.4.11 on a Supermicro SuperServer with Centos 5, and getting the following error trail on make. Googling the issue has found one user who tried: seems that commenting out typedef int bool; in xpp/xdefs.h on line 93 works that out, but don't know if it's completely right thing to do Roman http://bugs.digium.com/12889 (No new details there at the time of writing this) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
LOL, I agree, it _did_ sound a little complicated than to just schedule a call in the future. I apologize for not being able to find this on the wiki earlier when I searched. The other cron jobs and everything probably bring _something_ to the table. I wonder what. Either way, please keep 'em coming boys, and yes I'd like to know the answer to Tzafrir's question about performance. There will probably be 1500 calls that need to be made over a span of 15 hours. Some might happen as a handful, some might not. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: June 19, 2008 4:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Website callback On Wed, 18 Jun 2008, Mark Hamilton wrote: Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as to exactly what time Asterisk will need to call. Then, Asterisk calls it (by way of call files? Either putting the call file in at the time it needs to be called, or I don't know what else) and then if the call is has a human on it, plays a message saying We're now transferring you to an agent. Please wait. And transfer that call to a queue. How can I do this? Is there something prebuilt like this? Reading the replies so-far... Cron jobs, databases, shell scripts... Ye Gods... Try reading the manual (or at least the wiki) http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Scroll down to the bit headed: How to schedule a Call in the Future Assuming you already have some PHP to write the call-file and move it into place, inserting a touch call after writing the file and moving it into place ought to be trivial... http://uk.php.net/manual/en/function.touch.php Gordon ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote: LOL, I agree, it _did_ sound a little complicated than to just schedule a call in the future. I apologize for not being able to find this on the wiki earlier when I searched. The other cron jobs and everything probably bring _something_ to the table. I wonder what. Either way, please keep 'em coming boys, and yes I'd like to know the answer to Tzafrir's question about performance. One very big benefit of using a database with cron jobs is that your web application does not need to run as the same user (or otherwise weaken security permissions) as the Asterisk daemon. If running as the same user, you'd have to either set both daemons to the same group (which means the web server has access to all other files that Asterisk writes) or world writable, which is even worse. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream Busy Light Fields
Hello Gordon,
On Thu, 19 Jun 2008, Jan Prunk wrote:
Hello !
I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18
The things work up to 80%, I can transfer the call by BLF button and I can
see the green (free) status and red (busy) status.
Firstly, make sure the GS phones are of a relatively new hardware revision
and have the latest firmware loaded...
Program-- 1.1.6.16Bootloader-- 1.1.6.5
is a current good version for GXP phones.
Yes we got all phones updated with newest firmware.
Here the output from extensions.conf:
[buster]
; the usual dialplan+extensions are here, then comes:
; Pickup BLF
exten = _**6,1,Pickup(${EXTEN:1})
exten = _**6,2,Hangup
This is wrong - you need :2 to skip over the 2 stars.
exten = _**11,1,Pickup(${EXTEN:2})
exten = _**11,2,Hangup
exten = _**6X,1,Pickup(${EXTEN:2})
exten = _**6X,2,Hangup
exten = _**7X,1,Pickup(${EXTEN:2})
exten = _**7X,2,Hangup
exten = _**8X,1,Pickup(${EXTEN:2})
exten = _**8X,2,Hangup
You might want to try:
exten = _**.,1,Pickup(${EXTEN:2})
exten = _**.,n,Hangup()
Ok I have tried adding these 2 lines, and the error which I get when calling
01 5863165, which then rings extension 65, and I try to accept the call on
extension 70 by a BLF button. It gives me error code.
-- Accepting overlap voice call from '015852977' to '5863165' on channel
0/1, span 3
-- Starting simple switch on 'Zap/7-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/7-1, SIP/65|17|rtk) in new
stack
Extension Changed 65[BLF] new state Ringing for Notify User 70
-- Called 65
-- SIP/65-081fb370 is ringing
-- Executing [EMAIL PROTECTED]:1] PickUp(SIP/70-b5f18268, 65) in new
stack
[2008-06-19 15:13:33] WARNING[7287]: channel.c:4347 ast_get_group: Ignoring
invalid group 65 (maximum group is 63)
-- No channel found 0.
== Spawn extension (buster, **65, 1) exited non-zero on 'SIP/70-b5f18268'
-- Channel 0/1, span 3 got hangup request, cause 16
Extension Changed 65[BLF] new state Idle for Notify User 70
== Spawn extension (buster, 5863165, 1) exited non-zero on 'Zap/7-1'
-- Hungup 'Zap/7-1'
Regards,
Jan
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Re: [asterisk-users] Website callback
On Thu, Jun 19, 2008 at 08:05:59AM -0500, Tilghman Lesher wrote: On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote: LOL, I agree, it _did_ sound a little complicated than to just schedule a call in the future. I apologize for not being able to find this on the wiki earlier when I searched. The other cron jobs and everything probably bring _something_ to the table. I wonder what. Either way, please keep 'em coming boys, and yes I'd like to know the answer to Tzafrir's question about performance. Test it yourself? for i in `seq 1500`; do something to create a call file sleep a_bit done One very big benefit of using a database with cron jobs is that your web application does not need to run as the same user (or otherwise weaken security permissions) as the Asterisk daemon. If running as the same user, you'd have to either set both daemons to the same group (which means the web server has access to all other files that Asterisk writes) or world writable, which is even worse. In any version you'll still need something with permissions to originate calls on Asterisk. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Website callback
On Thu, Jun 19, 2008 at 9:57 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Thu, Jun 19, 2008 at 08:05:59AM -0500, Tilghman Lesher wrote: On Thursday 19 June 2008 07:57:07 Mark Hamilton wrote: LOL, I agree, it _did_ sound a little complicated than to just schedule a call in the future. I apologize for not being able to find this on the wiki earlier when I searched. The other cron jobs and everything probably bring _something_ to the table. I wonder what. Either way, please keep 'em coming boys, and yes I'd like to know the answer to Tzafrir's question about performance. Test it yourself? for i in `seq 1500`; do something to create a call file sleep a_bit done One very big benefit of using a database with cron jobs is that your web application does not need to run as the same user (or otherwise weaken security permissions) as the Asterisk daemon. If running as the same user, you'd have to either set both daemons to the same group (which means the web server has access to all other files that Asterisk writes) or world writable, which is even worse. In any version you'll still need something with permissions to originate calls on Asterisk. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir I have done hundreds at once, takes a few seconds to handle (all SIP) Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Adding ;password=foo;method=bar to SIP uri
Is there any reason that the SIP INVITE URL shouldn't conform to the same syntax as RFC3986 standard URLs ( http://en.wikipedia.org/wiki/URI_scheme#Generic_syntax ), as specific to SIP according to RFCs 3969 and 3261? That would be, according to sip:user[:password]@host[:port][;uri-parameters][?headers] examples: sip:[EMAIL PROTECTED]priority=urgent sip:+1-212-555-1212:[EMAIL PROTECTED];user=phone Like sip:xyz:[EMAIL PROTECTED];Authorization=bar+realm%3Dbaz OR sip:xyz:[EMAIL PROTECTED];?Authorization:+bar;realm%3Dbaz or something along those lines, as per http://tools.ietf.org/html/rfc3261#page-194 ? On Thu, 2008-06-19 at 03:38 -0500, [EMAIL PROTECTED] wrote: Date: Wed, 18 Jun 2008 18:34:15 -0400 From: Tom Browning [EMAIL PROTECTED] Subject: [asterisk-users] Adding ;password=foo;method=bar to SIP uri To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 To send calls into a custom SER implementation, I need to be able to add some items to the URI that Asterisk will then send as part of the INVITE Asterisk dial SIP/[EMAIL PROTECTED] needs to become Asterisk dial SIP/[EMAIL PROTECTED];password=foo;method=bar This is not a registration password. It is a passsword associated with the destination xyz at location abc.com Asterisk 1.4.18.1 seems to glue the data as part of the hostname and fail to lookup abc.com Is there a way to manipulate the URI that will be sent in the INVITE to accomplish this? Thanks in advance, Tom -- (C) Matthew Rubenstein ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR for callee (called party)
Hi Asterisk Users,
my apologizes for cross posting.
I'm trying to make the next scenario in Asterisk DialPlan: Alice calls Bob,
Asterisk executes Dial application with G(context^exten^pri), after that Bob
answers the call, Asterisk transfers Alice to pri, Bob to pri+1. It should
be possible for example that in that context Asterisk executes different
scenarios for Bob and Alice and then connects Alice to Bob to let them
communicate. The problem is that I can not connect both sides for
conversation, Asterisk just hangs up after executes the scenarios.
*[AnswerPrompt]
exten = s,1,Goto(10)
exten = s,2,Playback(Announce1)
exten = s,10,Playback(Announce2)
[call-number]
exten = _X.,1,Dial(SIP/${EXTEN}|G(AnswerPrompt^s^1))
exten = _X.,n,Hangup()
*
Is there any solutions? Any help will be appropriate.
On Tue, May 20, 2008 at 3:56 PM, Alexander Olekhnovich
[EMAIL PROTECTED] wrote:
Thanks a lot, that's the answer i could dream of :)
On Tue, May 20, 2008 at 3:34 PM, Tony Mountifield
[EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Alexander Olekhnovich [EMAIL PROTECTED] wrote:
Could anyone please answer my question.
I want to make the next scenario be possible.
1. Caller call another user.
2. Callee (called party) picks up and enters IVR menu. And then
depending on
his choice he has variants to: transfer the call to another user,
transfer
to voicemail, answer, hangup, etc...
The problem is in the second part. As I remember Asterisk has an A(x)
parameter of Dial to play the Announce to callee, but is there a
possibility
to organize IVR for callee after he picks up the receiver?
Any help will be appropriate.
Use the G(context^ext^pri) option to Dial. This will transfer both the
calling and called parties into the dialplan when the call is answered.
Note that the calling party goes to priority pri and the called party
goes to priority pri+1, so that you can do different things for each.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Best Regards
Alexander Olekhnovich
--
Best Regards
Alexander Olekhnovich
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Re: [asterisk-users] IVR for callee (called party)
In article [EMAIL PROTECTED],
Alexander Olekhnovich [EMAIL PROTECTED] wrote:
I'm trying to make the next scenario in Asterisk DialPlan: Alice calls Bob,
Asterisk executes Dial application with G(context^exten^pri), after that Bob
answers the call, Asterisk transfers Alice to pri, Bob to pri+1. It should
be possible for example that in that context Asterisk executes different
scenarios for Bob and Alice and then connects Alice to Bob to let them
communicate. The problem is that I can not connect both sides for
conversation, Asterisk just hangs up after executes the scenarios.
*[AnswerPrompt]
exten = s,1,Goto(10)
exten = s,2,Playback(Announce1)
exten = s,10,Playback(Announce2)
[call-number]
exten = _X.,1,Dial(SIP/${EXTEN}|G(AnswerPrompt^s^1))
exten = _X.,n,Hangup()
*
Is there any solutions? Any help will be appropriate.
In most versions of Asterisk, the best you can do is to put both calls
into a Meetme room with a unique room number. The drawback with that is
that when one of the parties hangs up, it doesn't automatically hang up
the other party.
There have been one or two enhancements proposed in the past to allow
one channel to grab another and bridge to it, but I don't think such an
application has made it into official versions yet (1.4 or trunk).
Cheers
Tony
--
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] Grandstream Busy Light Fields
On Thu, 19 Jun 2008, Jan Prunk wrote:
You might want to try:
exten = _**.,1,Pickup(${EXTEN:2})
exten = _**.,n,Hangup()
Ok I have tried adding these 2 lines, and the error which I get when calling
01 5863165, which then rings extension 65, and I try to accept the call on
extension 70 by a BLF button. It gives me error code.
-- Accepting overlap voice call from '015852977' to '5863165' on channel
0/1, span 3
-- Starting simple switch on 'Zap/7-1'
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/7-1, SIP/65|17|rtk) in new
stack
Extension Changed 65[BLF] new state Ringing for Notify User 70
-- Called 65
-- SIP/65-081fb370 is ringing
-- Executing [EMAIL PROTECTED]:1] PickUp(SIP/70-b5f18268, 65) in new
stack
[2008-06-19 15:13:33] WARNING[7287]: channel.c:4347 ast_get_group: Ignoring
invalid group 65 (maximum group is 63)
-- No channel found 0.
== Spawn extension (buster, **65, 1) exited non-zero on 'SIP/70-b5f18268'
-- Channel 0/1, span 3 got hangup request, cause 16
Extension Changed 65[BLF] new state Idle for Notify User 70
== Spawn extension (buster, 5863165, 1) exited non-zero on 'Zap/7-1'
-- Hungup 'Zap/7-1'
Er, I don't get quite the same output as you - I'm on 1.2 though. A test
call I've just done - extension 109 called extension 100, and extension
101 (a grandstream phone) picked it up by pushing the BLF key
corresponding to extension 100:
-- Executing Dial(SIP/109-0820a178, IAX2/100SIP/100||WwTton) in new
stack
-- Called 100
-- SIP/100-081fe780 is ringing
Extension Changed 100 new state Ringing for Notify User 101
-- Executing Pickup(SIP/101-081edf38, 100) in new stack
-- Executing Hangup(SIP/101-081edf38, ) in new stack
== Spawn extension (internal, **100, 2) exited non-zero on 'SIP/101-081edf38'
-- SIP/101-081edf38 answered SIP/109-0820a178
Extension Changed 100 new state Idle for Notify User 101
== Spawn extension (macro-dialInternal, s, 53) exited non-zero on
'SIP/109-0820a178' in macro 'dialInternal'
So your pickup is picking up a group - seems odd to me, but maybe the
behaviour changed after 1.2 ?
One other thing - do you have
exten = 65,1,Dial(SIP/65)
As pickup works on the extension not the channel... (ie. what do you dial
on a phone to make the SIP/65 ring? What does the DDI point to?)
Gordon
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[asterisk-users] Trouble with PRI config
I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
try pri_cpe instead of pri-cpe On Thursday 19 June 2008 12:51, Eve-Ellen Cole wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
Agreed. It looks like you've tried to tell the Avaya to be the network side but it doesn't seem to be acting like the network. Do what Steve suggested and see if you get a different result... -MC -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config pri_net usually when connecting to a legacy system. Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting Directory Behaviour (not)
On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote: Here are the details: If caller enters only three digits/letters: Jane Smith, Extension 123, If this is the person you are looking for... If the caller types in more than three letters, the person's name is not spoken, and the caller hears: Extension 123, If this is the person you are looking for... Callers, not hearing the person's name, have no idea if extension 123 is the correct extension and so are reluctant to confirm without hearing the person's name. What's with this? From the customer: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. The issue is that the 4th digit is actually interrupting the playback of the name, which is why they're not hearing it. Simple training issue. Or alternatively, you could play the name with DTMF-cut-through disabled, assuming that's not down inside C code... Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
This will happen if the other side is configured the same as the Asterisk side. i.e. PRI CPU mode on both ends or PRI NET mode on both ends. This can also happen if the line is in loopback mode at the far end. Eve-Ellen Cole wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CLI show queues NOT WORKING WELL
Just about 30 minutes that I can´t get real information from my Asterisk box. All agents seem to be available but is not true: QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet [EMAIL PROTECTED] asterisk]# asterisk -rx core show channels Channel Location State Application(Data) SIP/641-08cef808 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/641|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/65-1) Zap/65-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/64-1 [EMAIL PROTECTED]: Up (None) SIP/625-09766788 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/625|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/66-1) Zap/66-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_02|tT|||1800) SIP/620-09358088 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/620|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/63-1) Zap/63-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/94-1 (None) Up Bridged Call(SIP/623-b2b1d070) SIP/623-b2b1d070 [EMAIL PROTECTED]:3 Up Dial(Zap/g3/2714269||tTrRS) SIP/615-08a892c0 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Please help me with this issue! Thanks, ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + zap + sangoma A104D - how to setup call using particular timeslot
Hi all, I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten = TestTrakt,1,Dial(ZAP/1-2/517255333) exten = TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/sempron-b2ae1918, ZAP/1-2/517255333) in new stack [Jun 19 21:22:20] WARNING[23814]: chan_zap.c:7966 zt_request: Unknown option '-' in '1-2/517255333' -- Requested transfer capability: 0x00 - SPEECH -- Called 1-2/517255333 -- Hungup 'Zap/1-1' == Spawn extension (na-miasto, TestTrakt, 1) exited non-zero on 'SIP/sempron-b2ae1918' any idea how to force Asterisk to push call via particular timeslot? Regards, Marcin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
You'll probably need to turn on pri debugging for this span and then capture the output from when you connect the T1 cable. That might yield some clues, like whether or not any activity is happening on the d-channel and if so, if there are any errors that might tell you what's going on. -MC From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eve-Ellen Cole Sent: Thursday, June 19, 2008 12:39 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Trouble with PRI config Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config pri_net usually when connecting to a legacy system. Thanks, Steve T On Thu, Jun 19, 2008 at 1:38 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: The underscore helped, but didn't resolve the real issue. Now I get the following messages: [Jun 19 13:36:15] WARNING[4288] chan_zap.c: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. [Jun 19 13:36:16] WARNING[4288] chan_zap.c: No D-channels available! Using Primary channel 48 as D-channel anyway! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 19, 2008 1:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Trouble with PRI config Try underscore _ rather than dash - Thanks, Steve T On Thu, Jun 19, 2008 at 12:51 PM, Eve-Ellen Cole [EMAIL PROTECTED] wrote: I'm trying to connect Asterisk 1.4.20 to Avaya Definity G3R v11.1 via a T1 crossover, and I'm currently stuck. Anyone have any thoughts on what I can do to get past this? Asterisk side Digium TE220B w/ green LED (using port 2) Zaptel.conf span=2,1,0,esf,b8zs bchan=25-47 dchan=48 loadzone = us defaultzone=us Zapata.conf context=default switchtype=national ; T1 PRI to Avaya Definity G3R context=from_pbx signalling=pri-cpe group=3 channel = 25 Avaya side TN464GP Ds1 01C14 Framing mode: esf Line coding: b8zs Signaling mode: isdn-pri Connect: Network Protocol version: b (national) Near-end CSU type: other (for the T1 crossover) Signaling group 6 Primary d-channel set to 01C14 When I restart Asterisk, the following lines get logged to /var/log/asterisk/messages: [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Unknown signalling method 'pri-cpe' [Jun 19 12:41:37] ERROR[28093] chan_zap.c: Signalling must be specified before any channels are. If I change signaling method to pri-net: [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Unknown signalling method 'pri-net' [Jun 19 12:49:42] ERROR[28184] chan_zap.c: Signalling must be specified before any channels are. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP over TCP development in 1.6 branch?
List, Could anybody speak to the status of development in 1.6 branch? I know support for SIP over TCP is pretty new / experimental but it seems active development of it has slowed or stopped in recent months. Is that a correct statement? Is SIP over TCP more a community project or something headed from Digium? I only ask to get a pulse of its status; not harp or demand people to work on it. Like everybody else, we have some dependencies on SIP over TCP, and have a few bugs open against it. Personally, I would love to help develop or submit patches for the bugs but would need a mentor for that. Either way, just looking to get some more info about the development status of it. Thanks again, PB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + zap + sangoma A104D - how to setup call using particular timeslot
In article [EMAIL PROTECTED], Marcin J. Kowalczyk [EMAIL PROTECTED] wrote: I need to setup call using particular timeslot on one of E1's. I've looked into http://www.voip-info.org/wiki/index.php?page=Asterisk+Zap+channels and it says that: exten = TestTrakt,1,Dial(ZAP/1-2/517255333) exten = TestTrakt,2,hangup should work and force call setup via span 1 (port 1) but when I try setup call rasterisk says: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/sempron-b2ae1918, ZAP/1-2/517255333) in new stack [Jun 19 21:22:20] WARNING[23814]: chan_zap.c:7966 zt_request: Unknown option '-' in '1-2/517255333' -- Requested transfer capability: 0x00 - SPEECH -- Called 1-2/517255333 -- Hungup 'Zap/1-1' == Spawn extension (na-miasto, TestTrakt, 1) exited non-zero on 'SIP/sempron-b2ae1918' any idea how to force Asterisk to push call via particular timeslot? Not sure where you got the idea to use 1-2 as the channel number. Just use Zap/1/517255333 to call on channel 1, Zap/2/517255333 for channel 2, etc. The channels are as numbered in /etc/zaptel.conf and /etc/asterisk/zapata.conf Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
On Thu, 19 Jun 2008, Eve-Ellen Cole wrote: Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. It's not my area of expertise, but I have issues with T1 numbering between vendors -- what they said was 1, 2, 3, 4 turned out to be 4, 3, 2, 1. You might try swapping the T1s and see what happens. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting Directory Behaviour (not)
On Thursday 19 June 2008 13:38:05 Jay R. Ashworth wrote: On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. The issue is that the 4th digit is actually interrupting the playback of the name, which is why they're not hearing it. Simple training issue. Or alternatively, you could play the name with DTMF-cut-through disabled, assuming that's not down inside C code... That's a non-starter. Power users like to be able to interrupt prompts and press '1' immediately when they are sure that they've got the right person. Changing it now would be considered a regression to many. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interesting Directory Behaviour (not)
On Thu, Jun 19, 2008 at 03:49:01PM -0500, Tilghman Lesher wrote: On Thursday 19 June 2008 13:38:05 Jay R. Ashworth wrote: On Wed, Jun 18, 2008 at 05:27:04PM -0500, Tilghman Lesher wrote: Annoying that people aren't following the directions and only entering 3 digits, but we've had some high level meetings here with a string of clients coming through in an unusually compressed frequency. And I've had 5 complaints over 2 days that callers couldn't find Jane Smith. The issue is that the 4th digit is actually interrupting the playback of the name, which is why they're not hearing it. Simple training issue. Or alternatively, you could play the name with DTMF-cut-through disabled, assuming that's not down inside C code... That's a non-starter. Power users like to be able to interrupt prompts and press '1' immediately when they are sure that they've got the right person. Changing it now would be considered a regression to many. Or, alternatively, you could play the first $USERCONF milliseconds of the name with DTMF-cut-through disabled. This is akin to modal dialogs that pop up with the default button disabled for $LONGER-THAN-THE-AVERAGE-HUMAN-REACTION-TIME so that you don't keep on typing from whatever window you were previously in, and accidentally hit the Yes, delete all my files and kill my wife [OK] button by typing ENTER before you figure out what happened. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Joseph Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over TCP development in 1.6 branch?
On Thu, 2008-06-19 at 15:50 -0400, Paul Belanger wrote: List, Could anybody speak to the status of development in 1.6 branch? I know support for SIP over TCP is pretty new / experimental but it seems active development of it has slowed or stopped in recent months. Is that a correct statement? Is SIP over TCP more a community project or something headed from Digium? I only ask to get a pulse of its status; not harp or demand people to work on it. Like everybody else, we have some dependencies on SIP over TCP, and have a few bugs open against it. Personally, I would love to help develop or submit patches for the bugs but would need a mentor for that. Either way, just looking to get some more info about the development status of it. Can be brief about it: it just works! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI show queues NOT WORKING WELL
On Thu, Jun 19, 2008 at 10:06 PM, Chento Arohuanca [EMAIL PROTECTED] wrote: Just about 30 minutes that I can´t get real information from my Asterisk box. All agents seem to be available but is not true: QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet [EMAIL PROTECTED] asterisk]# asterisk -rx core show channels Channel Location State Application(Data) SIP/641-08cef808 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/641|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/65-1) Zap/65-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/64-1 [EMAIL PROTECTED]: Up (None) SIP/625-09766788 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/625|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/66-1) Zap/66-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_02|tT|||1800) SIP/620-09358088 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/620|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/63-1) Zap/63-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/94-1 (None) Up Bridged Call(SIP/623-b2b1d070) SIP/623-b2b1d070 [EMAIL PROTECTED]:3 Up Dial(Zap/g3/2714269||tTrRS) SIP/615-08a892c0 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Please help me with this issue! Local channels don't support state information in Asterisk 1.4. For that you either need to use 1.6 or backport of state_interface for 1.4. Then you have to set call-limit for peers, and specify state_interface device when logging in agents. For more information please search for asterisk queue state, as this has been discussed several times. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
On Thu, Jun 19, 2008 at 4:11 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Thu, 19 Jun 2008, Eve-Ellen Cole wrote: Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. It's not my area of expertise, but I have issues with T1 numbering between vendors -- what they said was 1, 2, 3, 4 turned out to be 4, 3, 2, 1. You might try swapping the T1s and see what happens. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I have actually done this with a Definity G3 but the cards were not PRI, I had to use EM_W. This doesn't make sense to me though. Primary d-channel set to 01C14 I memory servers me correctly (and it has been a couple years) the 01 means cabinet 1, C is the slot, and 14 is the port number. I would expect it to say 01C24 for the D chan. I could be completely wrong but it is something to try and would explain why your PRI chans are not coming up. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
The d-channel on the Avaya would be 01C1424. The rest of 01C14 would be the b-channels. - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 19, 2008 6:41:06 PM GMT -05:00 US/Canada Eastern Subject: Re: [asterisk-users] Trouble with PRI config On Thu, Jun 19, 2008 at 4:11 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Thu, 19 Jun 2008, Eve-Ellen Cole wrote: Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. It's not my area of expertise, but I have issues with T1 numbering between vendors -- what they said was 1, 2, 3, 4 turned out to be 4, 3, 2, 1. You might try swapping the T1s and see what happens. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I have actually done this with a Definity G3 but the cards were not PRI, I had to use EM_W. This doesn't make sense to me though. Primary d-channel set to 01C14 I memory servers me correctly (and it has been a couple years) the 01 means cabinet 1, C is the slot, and 14 is the port number. I would expect it to say 01C24 for the D chan. I could be completely wrong but it is something to try and would explain why your PRI chans are not coming up. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
On Thu, Jun 19, 2008 at 6:41 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Thu, Jun 19, 2008 at 4:11 PM, Steve Edwards [EMAIL PROTECTED] wrote: On Thu, 19 Jun 2008, Eve-Ellen Cole wrote: Right again, getting a little closer (babysteps) ... no alarms on either system, but when I check the pri status in the CLI, I get PRI span 2/0: Provisioned, Down, Active. I've searched for clues, but am not coming up with the next step. It's not my area of expertise, but I have issues with T1 numbering between vendors -- what they said was 1, 2, 3, 4 turned out to be 4, 3, 2, 1. You might try swapping the T1s and see what happens. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I have actually done this with a Definity G3 but the cards were not PRI, I had to use EM_W. This doesn't make sense to me though. Primary d-channel set to 01C14 I memory servers me correctly (and it has been a couple years) the 01 means cabinet 1, C is the slot, and 14 is the port number. I would expect it to say 01C24 for the D chan. I could be completely wrong but it is something to try and would explain why your PRI chans are not coming up. Thanks, Steve Totaro BTW, if that does turn out to be the issue, feel free to figure out how many hours you would have spent to figure it out and you can PayPal me your hourly rate multiplied by how much time I saved you. Definity support ain't cheap! ;-) Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with PRI config
Primary d-channel set to 01C14. Why doesn't it say 01C1424 then? On Thu, Jun 19, 2008 at 7:48 PM, Eve-Ellen [EMAIL PROTECTED] wrote: The d-channel on the Avaya would be 01C1424. The rest of 01C14 would be the b-channels. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mapping multimedia keys: pressed key not recognized
Wrong listsorry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: June 19, 2008 4:38 AM To: Asterisk Users List Subject: Re: [asterisk-users] Mapping multimedia keys: pressed key not recognized On Wed, Jun 18, 2008 at 08:21:06PM -0400, OCG Technical Support wrote: I've tried a few approaches to making the multimedia keys on my kbd play nice with myth, but all have lead to dead ends. One such dead end is to post this question to the Asteris Users mailing list, I guess :-( Wrong list? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can't make asterisk work...how to test?
All, I've put a new asterisk server at another location and can't seem to get it working. What's the best strategy to debug connections? I'm doing inbound SIP only and have installed the server in the same way as I did on my DEV server. Running an nmap on localhost shows the port listening: -- [asterisk]/ nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 21:12 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1476 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- [planet]/etc/asterisk nmap -sU localhost Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2008-06-19 20:11 CDT Interesting ports on localhost.localdomain (127.0.0.1): Not shown: 1484 closed ports PORT STATE SERVICE ... 5060/udp open|filtered sip ... -- Is there a command-line tool I can run that will attempt a SIP connection to a SIP server and provide some diagnostics about whether it could authenticate or even connect? -- Dante ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
