[asterisk-users] Undefined problem Asterisk problem
Hello folks,
It seems that i have a unique problem. So, we have distributed some cisco
spa303 phone and connected them to our asterisk box. We have also lots of
cisco 7911 phones connected Cisco CallManager. We integrated whole system.
But there is a problem let me illustrate it. In our campus we are at
headquarters, call it location A. And we have also some campus away from
us, call it B and C.
Now, in the centre there is absolute no problem. But B's and C's 7911 are
not able to call our spa303 all around. But our spa303 at A or B, C have no
problem and able to call all type of phones.
A - A, B, C (means at A we can call every phone from all type of phone)
B's,C's spa303 - A, B, C (means B's and C's spa303 can call every phone at
A, B, C)
B's, C's cisco 7911 - B's,C's and A's cisco 7911 (means there is no
comminucation problem between call manager's 7911. )
B's, C's cisco 7911 - A's, B's, C's spa303 (means B's and C's cisco
7911 can not call A's, B's, C's spa303. If one 7911 at B or C attempts to
call our spa303 connected our asterisk box, asterisk shows this logs)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [@from-sip-external:1] Answer(SIP/X.X.X.X-0e3e,
) in new stack
== Spawn extension (from-sip-external, , 1) exited non-zero on
'SIP/X.X.X.X-0e3e'
-- Executing [h@from-sip-external:1] Hangup(SIP/X.X.X.X-0e3e, )
in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on
'SIP/X.X.X.X-0e3e'
Our from-sip-external context is:
[from-sip-external]
exten = _,1,Answer()
same = n,Set(a=${ODBC_kontrol(${EXTEN})})
same = n,GotoIf($[${a}==1]?e:h)
same = n(e),Macro(Ozel-Ara,${EXTEN},$[CDR(src)])
same = n(h),Macro(Ara,${EXTEN})
Log says that first line of context is executed but after that i can not
even comment anything cos it does not continue to execution.
If we want we can force to our spa303 ring. But even so when i picked up
phone, channel gets lost. I hope you figure my problem out. Have you any
ideas about ? I appreciate it.
Regards,
Cem Celebi
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[asterisk-users] Which tool to edit custom reports from CDR and queues logs ?
Hi, I would like to edit reports showing how fast operator and users answer incoming calls. Users are spread over 6 locations, each with its own asterisk instance. Operator is on main site. Users have casual extension but operator logs as queue agent. I've read or/and tried Star2Billing's CDR-Stats and A2Billing, Asternic Call center Stats,. I'm wondering if using a BI tool such as Jasper Reports would be preferable. Which tool would you suggest to build custom reports ? Regards -- _ -- 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] Set Language for VoiceMailMain
Hello, how do I set the language for the VoiceMailMain()-command ? How do I set the language per voicemail-box ? Thanks, Jonas. -- _ -- 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] [OpenIMSCore-Users] Asterisk
Creating new subscription Sending to 10.199.74.5:6060 (no NAT) Found peer '720001' for '720001' from 10.199.74.5:6060 Looking for 720001 in default (domain open-ims.test) --- Transmitting (no NAT) to 10.199.74.5:6060 --- SIP/2.0 404 Not Found These are the errors I am getting on asterisk. On Fri, Jan 11, 2013 at 6:27 PM, isshed isshed@gmail.com wrote: Hey Franz, I have a little progress here. Instead of getting 403 now I am getting 404 not found. and also failed authntication error is no more coming. Seems failed authentication is solved but 404 not found is new error introduced? Could you please let me know how to proceed. I am attaching my latest logs. Thanks, Isshed On Fri, Jan 11, 2013 at 11:00 AM, isshed isshed@gmail.com wrote: What I get on asterisk is the following line. Failed to authenticate user 720001 sip:720...@open-ims.test;tag=1447049631447050637 for SUBSCRIBE On Fri, Jan 11, 2013 at 10:11 AM, isshed isshed@gmail.com wrote: Hi Franz, I am attaching a trace file. Please consider user 720001 with client ip 10.199.74.34. openims ip is 10.199.74.5(domain open-ims.test) and astersk server is 10.199.74.3. Thanks, Isshed On Fri, Jan 11, 2013 at 1:07 AM, Franz Edler franz.ed...@technikum-wien.at wrote: This depends on the behavior of Asterisk for the specific service. I don’t know the DND service. It must behave as a standard conforming AS. ** ** A tracefile could give a first view on the scenario. ** ** Br Fr ** ** ** ** *Von:* openimscore-users-boun...@lists.berlios.de [mailto: openimscore-users-boun...@lists.berlios.de] *Im Auftrag von *isshed *Gesendet:* Donnerstag, 10. Januar 2013 18:00 *An:* asterisk-users@lists.digium.com; openimscore-us...@lists.berlios.de *Betreff:* [OpenIMSCore-Users] Asterisk ** ** Hi All, Is it possible to register a sip client to openims and subscribe DND to asterisk? Here asterisk is used as AS for openims? Assist please. thanks, Isshed. -- _ -- 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] Undefined problem Asterisk problem
The problem was incompetible codec, thanks all.
2013/1/11 Onur Cem Çelebi occel...@gmail.com
Hello folks,
It seems that i have a unique problem. So, we have distributed some cisco
spa303 phone and connected them to our asterisk box. We have also lots of
cisco 7911 phones connected Cisco CallManager. We integrated whole system.
But there is a problem let me illustrate it. In our campus we are at
headquarters, call it location A. And we have also some campus away from
us, call it B and C.
Now, in the centre there is absolute no problem. But B's and C's 7911 are
not able to call our spa303 all around. But our spa303 at A or B, C have no
problem and able to call all type of phones.
A - A, B, C (means at A we can call every phone from all type of phone)
B's,C's spa303 - A, B, C (means B's and C's spa303 can call every phone
at A, B, C)
B's, C's cisco 7911 - B's,C's and A's cisco 7911 (means there is no
comminucation problem between call manager's 7911. )
B's, C's cisco 7911 - A's, B's, C's spa303 (means B's and C's cisco
7911 can not call A's, B's, C's spa303. If one 7911 at B or C attempts to
call our spa303 connected our asterisk box, asterisk shows this logs)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [@from-sip-external:1]
Answer(SIP/X.X.X.X-0e3e, ) in new stack
== Spawn extension (from-sip-external, , 1) exited non-zero on
'SIP/X.X.X.X-0e3e'
-- Executing [h@from-sip-external:1] Hangup(SIP/X.X.X.X-0e3e,
) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on
'SIP/X.X.X.X-0e3e'
Our from-sip-external context is:
[from-sip-external]
exten = _,1,Answer()
same = n,Set(a=${ODBC_kontrol(${EXTEN})})
same = n,GotoIf($[${a}==1]?e:h)
same = n(e),Macro(Ozel-Ara,${EXTEN},$[CDR(src)])
same = n(h),Macro(Ara,${EXTEN})
Log says that first line of context is executed but after that i can not
even comment anything cos it does not continue to execution.
If we want we can force to our spa303 ring. But even so when i picked up
phone, channel gets lost. I hope you figure my problem out. Have you any
ideas about ? I appreciate it.
Regards,
Cem Celebi
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Re: [asterisk-users] Set Language for VoiceMailMain
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Which tool to edit custom reports from CDR and queues logs ?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Friday, January 11, 2013 4:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Which tool to edit custom reports from CDR and queues logs ? Hi, I would like to edit reports showing how fast operator and users answer incoming calls. Users are spread over 6 locations, each with its own asterisk instance. Operator is on main site. Users have casual extension but operator logs as queue agent. I've read or/and tried Star2Billing's CDR-Stats and A2Billing, Asternic Call center Stats,. I'm wondering if using a BI tool such as Jasper Reports would be preferable. Which tool would you suggest to build custom reports ? Regards My exposure to these tools seems to indicate that they are all SQL/MYSQL based engines. To combine the data from six sites my inclination would be to use something like Crystal Reports or Perl to roll my own (or possibly even as low tech as Excel). -- _ -- 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] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox
and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted
to answer the phone in English, then do voicemails in different
languages, this should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) -- redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Set Language for VoiceMailMain
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will
influence the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have
the [zonemessage] section in voicemail.conf that could probably be
tweaked to change the language without dialplan changes. Also in
sip.conf you can set language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 9:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of
mailbox and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you
wanted to answer the phone in English, then do voicemails in
different languages, this should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) -- redundant since default
is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Set Language for VoiceMailMain
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know until
you try it on your box.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will influence
the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Set Language for VoiceMailMain
Well, I thought you had tried it and thus could tell it with 100% certainty.
Thanks for your help.
Jonas.
On 01/11/2013 04:16 PM, Danny Nicholas wrote:
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know
until you try it on your box.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 9:15 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will
influence the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you
have the [zonemessage] section in voicemail.conf that could
probably be tweaked to change the language without dialplan
changes. Also in sip.conf you can set language by peer so you
could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Friday, January 11, 2013 9:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of
mailbox and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you
wanted to answer the phone in English, then do voicemails in
different languages, this should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) -- redundant since
default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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_
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Re: [asterisk-users] Set Language for VoiceMailMain
Tried it just now and that is indeed the way it works (100% for me).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Well, I thought you had tried it and thus could tell it with 100% certainty.
Thanks for your help.
Jonas.
On 01/11/2013 04:16 PM, Danny Nicholas wrote:
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know until
you try it on your box.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will influence
the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 5:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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[asterisk-users] FW: Correct auth, but based on stale nonce received from
Hello, I have a 16 port FXS device for register analog phones. I see TTL (time tol ive for re-register) option in fxs menu and I have to chose a time between 10 to 7200 second. All ports going to unregister after the time what I choose. Im getting a registration failed message. In asterisk logs I see the Correct auth, but based on stale nonce received from message. Any idea? Emre -- _ -- 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 = sign and double == sign.What is the difference and when to use the two properly?
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so
If i had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same = n(true),Goto(main,s,1)
same= n(false), Hangup()
would this be saying the same thing as above then?
exten = _,answer()
same= n,Set($[${a}=1]?true:false] -- single equal sign
in essence wouldn't i get the same result ? im confused on the double and
single equal sign and when to use the difference of the two. Would i get
the same result in both these expressions?
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Re: [asterisk-users] Which tool to edit custom reports from CDR and queues logs ?
We are just delivering version 2 of our ADTransform data connector. It would allow your to read in your CDR files, manipulate them, validate them and put out JasperReports based on the data. It has a plug-in based workflow engine so that file transfers, input,transformation, validation, logging, reporting and automated report/file delivery steps can be added as required. Out of the box it supports CSV and Excel files for input and output and other connectors such as JBDC, OBDC, webservices, extraction using custom APIs can be added as plug-ins. It supports JasperReports out of the box. It supports FTP for getting and sending data and e-mail for sending reports, logs and data. I am guessing that you might want to suck in all of the CDRs, apply some mappings to add location or user specific information for reporting and then run a series of JasperReports to provide summary and detail reports, possibly extract data for billing/chargeback and the deliver the data and reports through e-mail to the appropriate recipients. It is a pure Java application that is OS agnostic. We are looking at Raspberry support as part of version 3. The original motivation for the product was the LMS market where data needs to be integrated from Payroll and HRIS to be feed into the LMS and certification and other training history data needs to be extracted to go to work scheduling or HRIS systems. As a small Asterisk user, I think that I can understand where your requirement is coming from. http://www.artifact-software.com/?page_id=929 is the website link if you want more info. A short brochure is available. If anyone wants one, please contact me off-list. Ron On 11/01/2013 5:22 AM, Olivier wrote: Hi, I would like to edit reports showing how fast operator and users answer incoming calls. Users are spread over 6 locations, each with its own asterisk instance. Operator is on main site. Users have casual extension but operator logs as queue agent. I've read or/and tried Star2Billing's CDR-Stats and A2Billing, Asternic Call center Stats,. I'm wondering if using a BI tool such as Jasper Reports would be preferable. Which tool would you suggest to build custom reports ? Regards -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] FW: Correct auth, but based on stale nonce received from
Hello, On 11.01.2013 17:34, Emre Özcan (Alfacom) wrote: I have a 16 port FXS device for register analog phones. I see TTL (time tol ive for re-register) option in fxs menu and I have to chose a time between 10 to 7200 second. All ports going to unregister after the time what I choose. Im getting a registration failed message. In asterisk logs I see the Correct auth, but based on stale nonce received from message. Try pedantic=no in sip.conf. Is the FXS device an Audiocodes? Have fun, -- Cristi -- _ -- 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 = sign and double == sign.What is the difference and when to use the two properly?
On Friday 11 January 2013, penguin wrote:
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so
If i had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same = n(true),Goto(main,s,1)
same= n(false), Hangup()
would this be saying the same thing as above then?
exten = _,answer()
same= n,Set($[${a}=1]?true:false] -- single equal sign
in essence wouldn't i get the same result ? im confused on the double and
single equal sign and when to use the difference of the two. Would i get
the same result in both these expressions?
Generally, one = sign means you're telling. Two == signs means you're asking.
It's amusing (for sadists) to see ex-BASIC programmers trip up over this and
write something like this:
if (denominator = 0) {
printf (Can't divide by zero!\n);
} else {
answer = numerator / denominator
};
This will never print Can't divide by zero! because you are actually
assigning a value to a variable right there in the conditional, and returning
the assigned value. Since this is zero, the if() will fail and drop through
to the else clause -- and then, just to confuse you, the program will crash
with Floating point exception anyway.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Single = sign and double == sign.What is the difference and when to use the two properly?
On 01/11/2013 12:20 PM, A J Stiles wrote:
I try to write comparisons as != where possible and then there is no
confusion and less mistakes possible.
Most compilers will warn on the example below now.
On Friday 11 January 2013, penguin wrote:
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so
If i had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same = n(true),Goto(main,s,1)
same= n(false), Hangup()
would this be saying the same thing as above then?
exten = _,answer()
same= n,Set($[${a}=1]?true:false] -- single equal sign
in essence wouldn't i get the same result ? im confused on the double and
single equal sign and when to use the difference of the two. Would i get
the same result in both these expressions?
Generally, one = sign means you're telling. Two == signs means you're asking.
It's amusing (for sadists) to see ex-BASIC programmers trip up over this and
write something like this:
if (denominator = 0) {
printf (Can't divide by zero!\n);
} else {
answer = numerator / denominator
};
This will never print Can't divide by zero! because you are actually
assigning a value to a variable right there in the conditional, and returning
the assigned value. Since this is zero, the if() will fail and drop through
to the else clause -- and then, just to confuse you, the program will crash
with Floating point exception anyway.
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Re: [asterisk-users] Single = sign and double == sign.What is the difference and when to use the two properly?
On Fri, 11 Jan 2013, jon pounder wrote: I try to write comparisons as != where possible and then there is no confusion and less mistakes possible. Most compilers will warn on the example below now. Or you can write comparisons as 'constant operator variable' like: if (0 == onhook) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Correct auth, but based on stale nonce received from
I only see that message when I have sip debug enabled. It appears harmless. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emre Özcan (Alfacom) Sent: Friday, January 11, 2013 10:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] FW: Correct auth, but based on stale nonce received from Hello, I have a 16 port FXS device for register analog phones. I see TTL (time tol ive for re-register) option in fxs menu and I have to chose a time between 10 to 7200 second. All ports going to unregister after the time what I choose. Im getting a registration failed message. In asterisk logs I see the Correct auth, but based on stale nonce received from message. Any idea? Emre -- _ -- 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 = sign and double == sign.What is the difference and when to use the two properly?
In Asterisk extensions.conf and extensions.ael inside $[] = and == are the
same comparison operator. I can't quote where I saw this, but it has been
documented somewhere. The == was added to make things more programmer
friendly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of penguin
Sent: Friday, January 11, 2013 10:50 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Single = sign and double == sign.What is the
difference and when to use the two properly?
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so If i
had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same = n(true),Goto(main,s,1)
same= n(false), Hangup()
would this be saying the same thing as above then?
exten = _,answer()
same= n,Set($[${a}=1]?true:false] -- single equal sign
in essence wouldn't i get the same result ? im confused on the double and
single equal sign and when to use the difference of the two. Would i get the
same result in both these expressions?
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[asterisk-users] How often to restart Asterisk...
Had my Asterisk instance stop responding to incoming/outgoing calls today. Had to kill -9 the asterisk process and restart it to get it back. Not really looking for help on that as the instance is version 1.6 and is due to be replaced with an upgraded version shortly. However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until there is a problem? This box had 119 days of up time on the Asterisk process. I have a client that I installed an Elastix instance on and the last time I checked it, it was up to almost 500 days of up time without an asterisk restart. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208-- _ -- 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] How often to restart Asterisk...
The general rule seems to be, don't restart it unless there's a problem or you hear of memory leaks. I had a version of 1.4 that I restarted every night because I read about memory leaks, but I hear of 1.2 installs that have been running continuously for 10 years. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Larsen Sent: Friday, January 11, 2013 3:07 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How often to restart Asterisk... Had my Asterisk instance stop responding to incoming/outgoing calls today. Had to kill -9 the asterisk process and restart it to get it back. Not really looking for help on that as the instance is version 1.6 and is due to be replaced with an upgraded version shortly. However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until there is a problem? This box had 119 days of up time on the Asterisk process. I have a client that I installed an Elastix instance on and the last time I checked it, it was up to almost 500 days of up time without an asterisk restart. Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 -- _ -- 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] How often to restart Asterisk...
On Fri, Jan 11, 2013 at 2:06 PM, Kevin Larsen kevin.lar...@pioneerballoon.com wrote: However, this does make me wonder, do you restart periodically to try to avoid issues or do you just let things run until there is a problem? This box had 119 days of up time on the Asterisk process. I have a client that I installed an Elastix instance on and the last time I checked it, it was up to almost 500 days of up time without an asterisk restart. I've had boxes run for years, and others have problems in a month or two. I have a general practice of having a reboot cron job on critical servers at 3am on Sunday. Our customer SLA allows for a maintenance period during this time. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Outoing Calls Motif Google Voice Calls Ring After Pick-up
Hey everyone, I just put in a fix for the underlying issue that was causing this to occur. It will be out in a future Asterisk 11 release. If you want the change now and are comfortable using patch you can retrieve the diff at: http://svnview.digium.com/svn/asterisk/branches/11/res/res_xmpp.c?r1=378411r2=378917view=patch and apply it against your Asterisk 11 source code. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up
File, thanks for that quick fix! Using it now. -- kuj On Fri, Jan 11, 2013 at 4:09 PM, Joshua Colp jc...@digium.com wrote: Hey everyone, I just put in a fix for the underlying issue that was causing this to occur. It will be out in a future Asterisk 11 release. If you want the change now and are comfortable using patch you can retrieve the diff at: http://svnview.digium.com/svn/**asterisk/branches/11/res/res_** xmpp.c?r1=378411r2=378917**view=patchhttp://svnview.digium.com/svn/asterisk/branches/11/res/res_xmpp.c?r1=378411r2=378917view=patchand apply it against your Asterisk 11 source code. 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
