[asterisk-users] addressing peers dynamically
hi, in my small setup (just for home usage) i have 5 phones configured. but only 2 of them are permanent connected to asterisk. nevertheless i want to address beside those two phones other peers if available. nowadays i address them always, resulting in error messages: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) Is there a way to avoid those messages? I think about something like a virtual queue (please excuse if the wording is incorrect, i am not in too deep to asterisk, i am more firm with genesys) that is addressed and peers are registering to that queue. is that the right path, or am i barking the wrong tree? regards, andre -- _ -- 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] Allowing peers from specific subnet only
Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing peers from specific subnet only
Hi You can achieve this with either permit/deny or contactpermit/contactdeny Single IP should be defined like : deny=0.0.0.0/0.0.0.0 permit=192.168.2.1/255.255.255.255 And networks in similar way with appropriate subnet mask deny=0.0.0.0/0.0.0.0 permit=192.168.2.0/255.255.255.0 You can also specify multiple subnets with ';' like: permit=192.168.2.0/255.255.255.0;192.168.1.0/255.255.255.0 Regards, Zohair Raza On Mon, Nov 19, 2012 at 4:12 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Allowing peers from specific subnet only
bilal ghayyad wrote: Hi; Hola, How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? This can be accomplished using ACLs. They are configured using the deny and permit settings within sip.conf. Example: deny=0.0.0.0/0.0.0.0 permit=172.16.10.0/255.255.255.0 This permits only devices from the 172.16.10.1-172.16.10.255 range. For cases where you may want to configure this in one place and share it around Asterisk 11 has introduced what are called Named ACLs. You can find further information on those at https://wiki.asterisk.org/wiki/display/AST/Named+ACLs 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] addressing peers dynamically
Andre Gronwald wrote: hi, Hola, in my small setup (just for home usage) i have 5 phones configured. but only 2 of them are permanent connected to asterisk. nevertheless i want to address beside those two phones other peers if available. nowadays i address them always, resulting in error messages: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) Is there a way to avoid those messages? I think about something like a virtual queue (please excuse if the wording is incorrect, i am not in too deep to asterisk, i am more firm with genesys) that is addressed and peers are registering to that queue. is that the right path, or am i barking the wrong tree? Is there any particular reason you don't want to do this or is it just because you get the Unable to create channel message? There's nothing really *wrong* with that message in your case. 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] meetme race condition
Jerry Geis wrote: I think I have a race condition. I am running something like this in my dialplan call agi to bring my list of devices into my MeetMe Playback beep start MeetMe() So in fact the meetme is not started before I bring the list of devices into the meetme. How can I do this differently so the MeetMe is started first or how can I wait in my AGI on the MeetMe to start because the MeetMe wont start until I exit the AGI... - or how do I in the dialplan wait for for the Meetme because I do have a stage where I redirect the Call into the MeetMe. so how do I inject a line that waits there for the MeetMe to be active??? Can you clarify what you mean by MeetMe to be active? What MeetMe options are you using and what is your configuration like? With the proper combination of options it shouldn't matter who gets into the conference bridge first. This is what Page essentially does, with the difference being that only the channel executing Page() can talk. If that behavior is what you are trying to accomplish I suggest you use that instead. 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] Noise on phones while speaking...
Carlos Chavez wrote: The card itself does not have hardware echo cancellation so we use MG2. I am not fixated on the card because this should not affect a SIP to SIP internal call unless the card is really defective and provides bad timing to Asterisk. Actually when bridging channels Asterisk acts as either a low level packet router (Packet2Packet or Local bridge - RTP packet is read in, minimally modified, and immediately sent back out) or as a higher level media forwarder (RTP packet is read in, dissected some, stuffed into internal data structure, shipped off to other channel, RTP header added, packet sent - although monitoring/recording/transcoding is involved it's in that list of operations too). Timing from an external source isn't used. So really, I'm fairly certain it's something to do with your phones. If you could post a short snippet of a phone calling another and the bridge that occurs I could be more certain. 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] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Face wrote: Hello, Hola, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603@DLPN_AlDimnaDialPlan:601] Dial(SIP/601-0002, SIP/603) in new stack [Nov 16 06:42:33] WARNING[15547][C-0004]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/601-0002' status is 'CHANUNAVAIL' and would not go to voicemail? Unfortunately without more information (dialplan involved, complete console output, sip show peer 603) it's impossible to fathom any potential reason why this is occurring. I suspect that's why nobody has responded to you until now. If you can provide that information I'm sure we can all help to determine if there really is an issue at work here! 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] Allowing peers from specific subnet only
Hello In SIP.find you can to use Deny=0.0.0.0/0.0.0.0 Permit=192.168.1.25/255.255.255 Regards On Nov 19, 2012 7:12 AM, bilal ghayyad bilmar...@yahoo.com wrote: Hi; How I can make my configuration to allow the sip phones only from specific IP addresses range (for example from 192.168.10.1 - 192.168.10.50) to be allowed to connect for asterisk? In other words, in addition to be authenticated based on the username and password, it is required that the IP address of the Phone to be from this range. How? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] meetme race condition
Can you clarify what you mean by MeetMe to be active? What MeetMe options are you using and what is your configuration like? With the proper combination of options it shouldn't matter who gets into the conference bridge first. This is what Page essentially does, with the difference being that only the channel executing Page() can talk. If that behavior is what you are trying to accomplish I suggest you use that instead. Josh Well I'm not sure whats happening then. I run the situation and test 100 times, my seven devices (all running asterisk 1.4.43) 99.9% of the time join the conference. One in 100 times- one of my seven devices did not join the conference. I am trying to figure out why. Jerry -- _ -- 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] If would possible use a custom function in Asterisk Dialplan
Hello,
If would be possible to use a function concept in side of Asterisk DialPlan
For example:
I have following logic in my dial plan remove country code a add an 0 before
the rest of the numbers
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN}))
; remove my country code
exten = _X.,n,
GotoIf($[${CALLERID(num):0:4}=${country-code}]?international-format:national-format)
exten = _X.,n(international-format), Set(CALLERID(num)=0${CALLERID(num):4})
exten = _X.,n(national-format), NoOp(call ID: ${CALLERID(num)} exten:
${EXTEN}))
Do you think if would be possible that I could write a function something like
REMOVEMYCOUNTRYCODE(${NUM}) with a return value of a number with out country
code and with an 0 add in front of the rest of the numbers.
like
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN}))
; remove my county code
exten = _X.n, Set(CALLERID(num=REMOVEMYCOUNTRYCODE(${CALLERID(num)} ));
then I have to define this function in someway ……
I am trying to googling for a while but I did not find any idea to achieve this
task.
I would appreciate if someone have an idea…
Thanks for your time in advance.
longst
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Re: [asterisk-users] If would possible use a custom function in Asterisk Dialplan
You could do it as a function if you are C literate. The simpler way would
be to do it as an AGI where you passed the ${EXTEN} value to the AGI and had
the AGI pass the modified number back as a dialplan variable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shitian Long
Sent: Monday, November 19, 2012 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] If would possible use a custom function in
Asterisk Dialplan
Hello,
If would be possible to use a function concept in side of Asterisk
DialPlan
For example:
I have following logic in my dial plan remove country code a add an 0
before the rest of the numbers
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN})) ; remove
my country code exten = _X.,n,
GotoIf($[${CALLERID(num):0:4}=${country-code}]?international-format:nati
onal-format)
exten = _X.,n(international-format), Set(CALLERID(num)=0${CALLERID(num):4})
exten = _X.,n(national-format), NoOp(call ID: ${CALLERID(num)} exten:
${EXTEN}))
Do you think if would be possible that I could write a function something
like REMOVEMYCOUNTRYCODE(${NUM}) with a return value of a number with out
country code and with an 0 add in front of the rest of the numbers.
like
exten = _X.,1, NoOp( call ID ${CALLERID(num)} exten: ${EXTEN})) ; remove
my county code exten = _X.n,
Set(CALLERID(num=REMOVEMYCOUNTRYCODE(${CALLERID(num)} ));
then I have to define this function in someway ..
I am trying to googling for a while but I did not find any idea to achieve
this task.
I would appreciate if someone have an idea.
Thanks for your time in advance.
longst
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Re: [asterisk-users] addressing peers dynamically
I had a similar problem (I work on 3 lans; when my firewall is down, the two
non-native lans are unaccessible) I wrote an AGI to execute sip show
peers and process only the ones that return OK and pass my peer numbers to
the AGI like this -
[dialall]
Exten = s,1,AGI(sipcheck.agi,100,200,300)
exten = s,n,Gotoif($[ ${LEN(${DIAL-100})} != 7]?dialall,s,4)
exten = s,n,Set(TODIAL=${TODIAL}''${DIAL-100})
exten = s,n,Gotoif($[ ${LEN(${DIAL-200})} != 7]?dialall,s,6)
exten = s,n,Set(TODIAL=${TODIAL}''${DIAL-200})
exten = s,n,Gotoif($[ ${LEN(${DIAL-300})} != 7]?dialall,s,8)
exten = s,n,Set(TODIAL=${TODIAL}''${DIAL-300)
exten = s,n,Dial(${TODIAL},40,i,KktTm)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Gronwald
Sent: Monday, November 19, 2012 2:00 AM
To: asterisk-users
Subject: [asterisk-users] addressing peers dynamically
hi,
in my small setup (just for home usage) i have 5 phones configured. but only
2 of them are permanent connected to asterisk.
nevertheless i want to address beside those two phones other peers if
available. nowadays i address them always, resulting in error messages:
Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Is there a way to avoid those messages? I think about something like a
virtual queue (please excuse if the wording is incorrect, i am not in too
deep to asterisk, i am more firm with genesys) that is addressed and peers
are registering to that queue.
is that the right path, or am i barking the wrong tree?
regards,
andre
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[asterisk-users] Max app_voicemail line length
We are getting this message on an Asterisk 1.4.44 box. [2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too long (1323). Truncating. I know Asterisk removed many of limitations in string lengths in in 1.6+. Does anyone know if this also applies to app_voicemail? -- _ -- 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] Max app_voicemail line length
I can tell you this warning does not exist in 10.9.0 or 11.0.0. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, November 19, 2012 12:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max app_voicemail line length We are getting this message on an Asterisk 1.4.44 box. [2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too long (1323). Truncating. I know Asterisk removed many of limitations in string lengths in in 1.6+. Does anyone know if this also applies to app_voicemail? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conf into a call in progress
On Sun, Nov 18, 2012 at 11:32 AM, Michael voip.quest...@gmail.com wrote: Gentlemen, So, from your answers I understand that I have 2 options: 1. AMI Redirect command 2. Asterisk command ChannelRedirect I'm inclined to prefer the 2nd option, as we've never used AMI, but I don't know if it can be web-initiated. If you're unfamiliar with the AMI, I would strongly suggest becoming familiar with it. We use PHP with a socket connection to the asterisk AMI and it works fantastically. This is precisely the kind of thing the AMI was meant to do. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- 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] Max app_voicemail line length
Thanks. We will never upgrade to Asterisk 10 and we won't be upgrading to Asterisk 11 for 12 - 18 months. In my experience with Asterisk 1.4, 1.6, and 1.8 is that it takes that long for Asterisk to be stable enough for our use. I'm STILL in therapy because of the if you receive a VM while you are checking VM your mailbox will be corrupted bug and that bug was fixed many months ago. I do NOT want to go thru THAT again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 19, 2012 1:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Max app_voicemail line length I can tell you this warning does not exist in 10.9.0 or 11.0.0. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, November 19, 2012 12:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max app_voicemail line length We are getting this message on an Asterisk 1.4.44 box. [2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too long (1323). Truncating. I know Asterisk removed many of limitations in string lengths in in 1.6+. Does anyone know if this also applies to app_voicemail? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Fwd: Errors Compiling Libpri-1.4.13
Good Day dear members, We are trying to test asterisk in our office to extend the reach of our present proprietary pabx system if successful. I am using an oracle virualbox 4.2.4 as the virtual server platform with ubuntu 12.04.1 server as the operating system. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. thanks Adolphus Enaboifo gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT prisched.o -MF .prisched.o.d -MP -c -o prisched.o prisched.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT q931.o -MF .q931.o.d -MP -c -o q931.o q931.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri_aoc.o -MF .pri_aoc.o.d -MP -c -o pri_aoc.o pri_aoc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri_cc.o -MF .pri_cc.o.d -MP -c -o pri_cc.o pri_cc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pri_facility.o -MF .pri_facility.o.d -MP -c -o pri_facility.o pri_facility.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT asn1_primitive.o -MF .asn1_primitive.o.d -MP -c -o asn1_primitive.o asn1_primitive.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose.o -MF .rose.o.d -MP -c -o rose.o rose.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_address.o -MF .rose_address.o.d -MP -c -o rose_address.o rose_address.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_aoc.o -MF .rose_etsi_aoc.o.d -MP -c -o rose_etsi_aoc.o rose_etsi_aoc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_cc.o -MF .rose_etsi_cc.o.d -MP -c -o rose_etsi_cc.o rose_etsi_cc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_diversion.o -MF .rose_etsi_diversion.o.d -MP -c -o rose_etsi_diversion.o rose_etsi_diversion.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_ect.o -MF .rose_etsi_ect.o.d -MP -c -o rose_etsi_ect.o rose_etsi_ect.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_etsi_mwi.o -MF .rose_etsi_mwi.o.d -MP -c -o rose_etsi_mwi.o rose_etsi_mwi.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_other.o -MF .rose_other.o.d -MP -c -o rose_other.o rose_other.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_q931.o -MF .rose_q931.o.d -MP -c -o rose_q931.o rose_q931.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_aoc.o -MF .rose_qsig_aoc.o.d -MP -c -o rose_qsig_aoc.o rose_qsig_aoc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_cc.o -MF .rose_qsig_cc.o.d -MP -c -o rose_qsig_cc.o rose_qsig_cc.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_ct.o -MF .rose_qsig_ct.o.d -MP -c -o rose_qsig_ct.o rose_qsig_ct.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_diversion.o -MF .rose_qsig_diversion.o.d -MP -c -o rose_qsig_diversion.o rose_qsig_diversion.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_mwi.o -MF .rose_qsig_mwi.o.d -MP -c -o rose_qsig_mwi.o rose_qsig_mwi.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT rose_qsig_name.o -MF .rose_qsig_name.o.d -MP -c -o rose_qsig_name.o rose_qsig_name.c gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT version.o -MF .version.o.d -MP -c -o version.o version.c ar rcs libpri.a copy_string.o pri.o q921.o prisched.o q931.o pri_aoc.o pri_cc.o pri_facility.o asn1_primitive.o rose.o rose_address.o rose_etsi_aoc.o rose_etsi_cc.o rose_etsi_diversion.o rose_etsi_ect.o rose_etsi_mwi.o rose_other.o rose_q931.o rose_qsig_aoc.o rose_qsig_cc.o rose_qsig_ct.o rose_qsig_diversion.o rose_qsig_mwi.o rose_qsig_name.o version.o ranlib libpri.a gcc -shared -Wl,-hlibpri.so.1.4 -o libpri.so.1.4 copy_string.o pri.o q921.o prisched.o q931.o pri_aoc.o pri_cc.o pri_facility.o asn1_primitive.o rose.o rose_address.o rose_etsi_aoc.o rose_etsi_cc.o rose_etsi_diversion.o rose_etsi_ect.o rose_etsi_mwi.o rose_other.o rose_q931.o rose_qsig_aoc.o rose_qsig_cc.o rose_qsig_ct.o rose_qsig_diversion.o
Re: [asterisk-users] Max app_voicemail line length
The warning is also non-existent in 1.8.17. I don't/won't mess with 1.6. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, November 19, 2012 12:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Max app_voicemail line length Thanks. We will never upgrade to Asterisk 10 and we won't be upgrading to Asterisk 11 for 12 - 18 months. In my experience with Asterisk 1.4, 1.6, and 1.8 is that it takes that long for Asterisk to be stable enough for our use. I'm STILL in therapy because of the if you receive a VM while you are checking VM your mailbox will be corrupted bug and that bug was fixed many months ago. I do NOT want to go thru THAT again. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, November 19, 2012 1:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Max app_voicemail line length I can tell you this warning does not exist in 10.9.0 or 11.0.0. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Monday, November 19, 2012 12:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Max app_voicemail line length We are getting this message on an Asterisk 1.4.44 box. [2012-11-19 08:49:27] WARNING[11785] app_voicemail.c: List of extensions is too long (1323). Truncating. I know Asterisk removed many of limitations in string lengths in in 1.6+. Does anyone know if this also applies to app_voicemail? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Errors Compiling Libpri-1.4.13
On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pridump.o -MF .pridump.o.d -MP -c -o pridump.o pridump.c pridump.c:45:24: fatal error: dahdi/user.h: No such file or directory compilation terminated. make: *** [pridump.o] Error 1 New in lipri 1.4.13 is a default dependency on DAHDI [1]. You should be good to go if you make sure that DAHDI is installed before compiling libpri. [1] http://svnview.digium.com/svn/libpri?view=revisionrevision=2294 Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel 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] addressing peers dynamically
Am 19.11.2012 19:00, schrieb asterisk-users-requ...@lists.digium.com: Subject: Re: [asterisk-users] addressing peers dynamically To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 50aa2586.80...@digium.com Content-Type: text/plain; charset=ISO-8859-1; format=flowed Andre Gronwald wrote: hi, Hola, in my small setup (just for home usage) i have 5 phones configured. but only 2 of them are permanent connected to asterisk. nevertheless i want to address beside those two phones other peers if available. nowadays i address them always, resulting in error messages: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) Is there a way to avoid those messages? I think about something like a virtual queue (please excuse if the wording is incorrect, i am not in too deep to asterisk, i am more firm with genesys) that is addressed and peers are registering to that queue. is that the right path, or am i barking the wrong tree? Is there any particular reason you don't want to do this or is it just because you get the Unable to create channel message? There's nothing really *wrong* with that message in your case. it is just because i think that something is not wrong (which is correct, because i address a currently not existing peer). and if there is a way to handle it better, then i would like to know it (virtual queues is just oversized, but maybe there is a simple usage of addressing only registered peers... regards, andre -- Andre Gronwald andregronwal...@gmail.com andre.gronw...@gmx.de PGP-0x9CDEE439 -- _ -- 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] Need advice on how to implement this ...
I need some advice on how to implement something in my dialplan. Here's the scenario. A call comes in on my [incoming] context and I answer it. The call turns out to be for my wife and she needs to answer it on a different handset somewhere else in the house. I've tried call parking but the wife acceptance factor is kind of low because we don't do it often enough for her to remember how to park and unpark. What I'd really like to do is define an easy DTMF sequence in features.conf (like 00) that would send the call back into my [incoming] context again, just like it was a new incoming call. Then it could be picked up anywhere in the house. What's the best way to go about this? I tried doing an AGI script that sets context/extension/priority to where I'd like for it to go but it doesn't seem to work. Am I on the right track or is there a better way to do this? -- Chris -- _ -- 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] Need advice on how to implement this ...
Have you looked into SLA? I have had good results with it. Will let asterisk act like a key system. Sent from Samsung tablet Chris Gentle gent...@gmail.com wrote: I need some advice on how to implement something in my dialplan. Here's the scenario. A call comes in on my [incoming] context and I answer it. The call turns out to be for my wife and she needs to answer it on a different handset somewhere else in the house. I've tried call parking but the wife acceptance factor is kind of low because we don't do it often enough for her to remember how to park and unpark. What I'd really like to do is define an easy DTMF sequence in features.conf (like 00) that would send the call back into my [incoming] context again, just like it was a new incoming call. Then it could be picked up anywhere in the house. What's the best way to go about this? I tried doing an AGI script that sets context/extension/priority to where I'd like for it to go but it doesn't seem to work. Am I on the right track or is there a better way to do this? -- Chris -- _ -- 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] Need advice on how to implement this ...
I need some advice on how to implement something in my dialplan. Here's the scenario. A call comes in on my [incoming] context and I answer it. The call turns out to be for my wife and she needs to answer it on a different handset somewhere else in the house. I've tried call parking but the wife acceptance factor is kind of low because we don't do it often enough for her to remember how to park and unpark. What I'd really like to do is define an easy DTMF sequence in features.conf (like 00) that would send the call back into my [incoming] context again, just like it was a new incoming call. Then it could be picked up anywhere in the house. What's the best way to go about this? I tried doing an AGI script that sets context/extension/priority to where I'd like for it to go but it doesn't seem to work. Am I on the right track or is there a better way to do this? You could use DTMF blind transfer to transfer the call back into the dialplan. Richard -- _ -- 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] Need advice on how to implement this ...
You can park the call, set the timeout low, and have it return to a ring group. On Nov 19, 2012 6:15 PM, Chris Gentle gent...@gmail.com wrote: I need some advice on how to implement something in my dialplan. Here's the scenario. A call comes in on my [incoming] context and I answer it. The call turns out to be for my wife and she needs to answer it on a different handset somewhere else in the house. I've tried call parking but the wife acceptance factor is kind of low because we don't do it often enough for her to remember how to park and unpark. What I'd really like to do is define an easy DTMF sequence in features.conf (like 00) that would send the call back into my [incoming] context again, just like it was a new incoming call. Then it could be picked up anywhere in the house. What's the best way to go about this? I tried doing an AGI script that sets context/extension/priority to where I'd like for it to go but it doesn't seem to work. Am I on the right track or is there a better way to do this? -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
On Mon, Nov 19, 2012 at 3:51 PM, Joshua Colp jc...@digium.com wrote: Face wrote: Hello, Hola, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603@DLPN_AlDimnaDialPlan:601] Dial(SIP/601-0002, SIP/603) in new stack [Nov 16 06:42:33] WARNING[15547][C-0004]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/601-0002' status is 'CHANUNAVAIL' and would not go to voicemail? Unfortunately without more information (dialplan involved, complete console output, sip show peer 603) it's impossible to fathom any potential reason why this is occurring. I suspect that's why nobody has responded to you until now. If you can provide that information I'm sure we can all help to determine if there really is an issue at work here! 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 Well, thanks for responding. I went back to 10.10.0 and things seem to be working fine now! -- Sincerely, falazemi -- _ -- 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] Need advice on how to implement this ...
On Mon, Nov 19, 2012 at 6:23 PM, Jared Baxley jared.bax...@gmail.comwrote: You can park the call, set the timeout low, and have it return to a ring group. Thanks to everyone for the suggestions. I decided to try this approach first and I think I have it working. However, I found a slight problem. According to the features.conf file, when a call returns to the comebackcontext, it will go to the s extension if the specific extension such as SIP_0004F2040001 does not exist. This does not seem to be the case. I've created an s extension in my [parkedcallstimeout] context but I can't make the returning call go to it. It fails with a message in the /var/log/asterisk/messages file: [2012-11-19 20:46:58] WARNING[23894][C-003e] pbx.c: Channel 'SIP/tcg-00c9' sent to invalid extension but no invalid handler: context,exten,priority=parkedcallstimeout,SIP_gigaset,1 To work around it, I had to define a specific SIP_ extension for each of my phones that might get sent to that context. Am I misunderstanding how this works? I'm running asterisk 11.0.1, so it could be a bug I suppose. Can anyone verify? -- Chris -- _ -- 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] tcptls ssl connection error
Hello All, Anyone have idea regarding below error. After applying all patch, still faced the same issue. -- Regards, Chandrakant Solanki On Fri, Nov 9, 2012 at 11:39 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello All, I am using asterisk 1.8.13.0 and which is running on TLS port and my request forwarded from opensips which is also run tls port. On both end my certificate is same. During search about this error, I found below blog and apply patch, then also found below error. https://issues.asterisk.org/jira/browse/ASTERISK-18345 https://issues.asterisk.org/jira/browse/ASTERISK-20559 Also applied r375023 [Nov 8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:37] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:37] WARNING[19274]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:39] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:39] WARNING[19356]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:49] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:49] WARNING[19357]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! -- Regards, Chandrakant Solanki -- _ -- 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
