[asterisk-users] Reffered By header is missing from SIP INVITE in call transfer scenarios
Hi, We are facing an issue with asterisk in the case of call-Transfer scenarios. Our requirement is to identify whether an incoming call is a fresh incoming call or a Transferred call from some other clients. We have a setup, where in the asterisk1.6 (as SIP server) is running in Linux machine, and three SIP clients(say A,B,C) registered to asterisk server are running in three different windows machines. With the above said setup, there is a call made from SIP client-A to SIP client-B through asterisk. The incoming call got answered in SIP client-B and transferred the call to SIP client-C via asterisk. Here the SIP client-B sends a REFER SIP message to Asterisk and a new INVITE (corresponds to the REFER SIP) is sent to SIP client-C. But there is no REFFERED BY Header added in the INVITE SIP message which is sent to SIP client-C. Due to this we are not able to identify the incoming call as Transferred call. So, we have two questions: 1) Are there any configuration changes in Asterisk to solve this (so that the asterisk handles the transfer in SIP signaling)? 2) Is there any other way in which we can identify a call as forwarded call? Best regards, Rajib -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem installing B410P BRI card for asterisk
Marco Did you get to the bottom of this. I've just come across the same problem today also with a B410P after upgrading from debian lenny( 2.6.26-2-686 ) to squeeze ( 2.6.32-5-686 ). On reboot I often get BUG: soft lockup - CPU#0 stuck for 61s[swapper:0] The only fix is to power off but here's the clue Feb 21 21:15:08 astrid kernel: [4.638977] hfc_multi :07:01.0: PCI INT A - GSI 19 (level, low) - IRQ 19 Feb 21 21:15:08 astrid kernel: [4.638982] Digium Inc. HFC-4S Card: defined at IOBASE 0xdc00 IRQ 19 HZ 250 leds-type 2 Feb 21 21:15:08 astrid kernel: [4.803576] wcb4xxp :07:01.0: PCI INT A - GSI 19 (level, low) - IRQ 19 Feb 21 21:15:08 astrid kernel: [4.803607] wcb4xxp :07:01.0: Identified Wildcard B410P (controller rev 1) at 0001dc00, IRQ 19 hfcmulti can be found here; /lib/modules/2.6.32-5-686/kernel/drivers/isdn/hardware/mISDN/hfcmulti.ko I've got the system running at the moment with the previous kernel, so nothing wrong with the B410P. I'm hoping that 'tomorrow' that adding the line below to /etc/modprobe.d/blacklist.conf will fix my lockups. blacklist hfcmulti Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind Sent: Friday, 30 December 2011 9:45 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Problem installing B410P BRI card for asterisk Dear all, I know this is more a Digium hardware than an Asterisk issue. Already posted a question at Digium, however also like to see whether anyone in the Asterisk community has encountered the following situation: I installed a Digium B410P BRI PCI card on my new asterisk server, following the steps specified in the manual. I can see the PCI card is available using the lspci command: ... 04:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 05:00.0 Ethernet controller [0200]: Intel Corporation 82574L Gigabit Network Connection [8086:10d3] 08:00.0 PCI bridge [0604]: ASPEED Technology, Inc. AST1150 PCI-to-PCI Bridge [1a03:1150] (rev 02) 09:00.0 VGA compatible controller [0300]: ASPEED Technology, Inc. ASPEED Graphics Family [1a03:2000] (rev 10) 0a:01.0 ISDN controller [0204]: Digium, Inc. Wildcard B410 quad-BRI card [d161:b410] (rev 01) ... I specified the following in my system.conf in /etc/dahdi: loadzone = nl defaultzone = nl span = 1,1,0,ccs,ami bchan = 1,2 hardhdlc = 3 I loaded the driver using sudo modprobe wcb4xxp. Next I ran dahdi_cfg -vv which returns: DAHDI Tools Version - 2.5.0.2 DAHDI Version: 2.5.0.2 Echo Canceller(s): HWEC Configuration == SPAN 1: CCS/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) Channel 02: Clear channel (Default) (Echo Canceler: none) (Slaves: 02) Channel 03: Hardware assisted D-channel (Default) (Echo Canceler: none) (Slaves: 03) 3 channels to configure. DAHDI_SPANCONFIG failed on span 1: No such device or address (6) I'm in doubt about the DAHDI_SPANCONFIG failed on span 1: No such device or address (6). Next, if i execute sudo dmesg as specified by the manual it returns a huge trace: [ 376.082907] Wrote 0x0 to register 0x1ab but got back 0x4 [ 376.594754] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.106605] Wrote 0x0 to register 0x1ab but got back 0x4 [ 377.618423] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.130266] Wrote 0x0 to register 0x1ab but got back 0x4 [ 378.642088] Wrote 0x0 to register 0x1ab but got back 0x4 [ 1202.812870] show_signal_msg: 21 callbacks suppressed [ 1202.812876] dahdi_tool[1277]: segfault at 3fc378fa0 ip 004021ac sp 7fff131dd930 error 4 in dahdi_tool[40+3000] And a lot of Wrote 0x0 to register 0x1ab but got back 0x4 statements. If i run dahdi_tools it fails with a segmentation fault. Any suggestions are appreciated! Kind regards, Marco Mooijekind. -- _ -- 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] how many UDP ports is required for 1 call
Hi, how many UDP ports is required for 1 call. and why . -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- 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 many UDP ports is required for 1 call
On 02/21/2012 07:30 AM, virendra bhati wrote: Hi, how many UDP ports is required for 1 call. and why . A 'call' is too ambiguous to answer your question. Is this a voice call, a video/voice call, a FAx call, a T.140 text call, or something else? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] how many UDP ports is required for 1 call
right now it's only voice call. But thanks for segregate the call. Now i want to know about all calls used port too. On Tue, Feb 21, 2012 at 7:06 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/21/2012 07:30 AM, virendra bhati wrote: Hi, how many UDP ports is required for 1 call. and why . A 'call' is too ambiguous to answer your question. Is this a voice call, a video/voice call, a FAx call, a T.140 text call, or something else? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- 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 many UDP ports is required for 1 call
As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. -- This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors. Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com On Feb 21, 2012, at 8:30 AM, virendra bhati virbh...@gmail.com wrote: Hi, how many UDP ports is required for 1 call. and why . -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- 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] how many UDP ports is required for 1 call
On 02/21/2012 07:51 AM, Alex Balashov wrote: As many ports as required by the nature of the call, i.e. the protocol(s) used for the bearer. For an IAX2 call, the answer is 'zero' for all of those call types (at least the ones that are supported in IAX2, not all of them are). For protocols that use RTP for media transport, two ports are required for each media stream (one for RTP, one for RTCP). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] asterisk-users@lists.digium.com Nacha Alert ID10416
asterisk-users@lists.digium.com wrote: Please click the link the NACHA site and update your user account:ID0664474 Interesting. Came from tipas...@gmail.com Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by 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] Park and PARKINGDYNAMIC
What release are you trying this with?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, February 20, 2012 5:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Park and PARKINGDYNAMIC
I have been trying to get the dynamic parking working.
For some reason when I park a call using this method the console says it is
using the default parking context not the one I am trying to specidfy. It
also is playing the parked extension to the caller. I am transfering the
call to an extension that is doing a goto to the context below. Any ideas
or examples on how to make this work. What I need to be able to do is have
multiple parking lots using the same extension pools but seperated by a
dynamic context of ${account}-Lot. So that each office suite cant cross
pickup another groups parked calls while using the same number pool of
110-120. I need the dynamic option as all of our calls are database driven
and we can't add a seperate entry per customer to the feautres.conf.
[MSIP-DynPark]
exten = s,1,NoOp(Dynamic Parking)
exten = s,n,NoOp(Return Parked Call)
exten = s,n,GoTo(${CUT(${l_ndeContext}-ndeArgs,~,1)},1)
exten = _XXX,1,Set(PARKINGDYNAMIC=parkinglot_small)
exten = _XXX,n,Set(PARKINGDYNEXTEN=110)
exten = _XXX,n,Set(PARKINGDYNPOS=111-120)
exten = _XXX,n,Set(PARKINGDYNCONTEXT=${account}-Lot)
;exten = _XXX,n,Set(PARKINGEXTEN=99)
exten = _XXX,n,Park()
[MSIP-DynParkPickup]
exten = _NXX,1,ParkedCall(${EXTEN},${account}-Lot)
exten = _NXX,hint,park:$EXTEN@${account}-Lot
Thanks
Bryant
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Re: [asterisk-users] Park and PARKINGDYNAMIC
Danny
I am on 1.8.x
I also have 1.10 boxes up but have not tried it there yet. According to the
change logs it should work from 1.8 and up but it does not appear to do so.
I have been going through the source code trying to figure it out as there
are no real doc's on it as of yet. If I can figure it out I want to put a
wiki page up so others don't have to go through the pains I am having with
it.
Thanks
Bryant
From: Danny Nicholas da...@debsinc.com
Sent: Tuesday, February 21, 2012 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Park and PARKINGDYNAMIC
What release are you trying this with? From:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Monday, February 20, 2012 5:34 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Park and PARKINGDYNAMIC I have been trying
to get the dynamic parking working.
For some reason when I park a call using this method the console says it is
using the default parking context not the one I am trying to specidfy. It
also is playing the parked extension to the caller. I am transfering the
call to an extension that is doing a goto to the context below. Any ideas
or examples on how to make this work. What I need to be able to do is have
multiple parking lots using the same extension pools but seperated by a
dynamic context of ${account}-Lot. So that each office suite cant cross
pickup another groups parked calls while using the same number pool of
110-120. I need the dynamic option as all of our calls are database driven
and we can't add a seperate entry per customer to the feautres.conf.
[MSIP-DynPark]
exten = s,1,NoOp(Dynamic Parking)
exten = s,n,NoOp(Return Parked Call)
exten = s,n,GoTo(${CUT(${l_ndeContext}-ndeArgs,~,1)},1)
exten = _XXX,1,Set(PARKINGDYNAMIC=parkinglot_small)
exten = _XXX,n,Set(PARKINGDYNEXTEN=110)
exten = _XXX,n,Set(PARKINGDYNPOS=111-120)
exten = _XXX,n,Set(PARKINGDYNCONTEXT=${account}-Lot)
;exten = _XXX,n,Set(PARKINGEXTEN=99)
exten = _XXX,n,Park()
[MSIP-DynParkPickup]
exten = _NXX,1,ParkedCall(${EXTEN},${account}-Lot)
exten = _NXX,hint,park:$EXTEN@${account}-Lot
Thanks
Bryant
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[asterisk-users] Define custom vm-login sound file per VM context?
Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? By default the sound file is vm-login.codec. Is there a way to specify which sound file is played per context or some other way to play a different sound file in place of vm-login? I have already replaced the default file and named it the same vm-login.x but still I am only able to play one file, not a different file depending on the VM context I send the caller to. I am sure someone has figured this out so, any shortcut to keep me from frying my brain on this would be appreciated. Thanks! --Todd -- _ -- 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] Define custom vm-login sound file per VM context?
I believe this is what you want. Instead of this Exten = _X.,123,Voicemail(100) Do Exten = _X.,123,playback(your-message) Exten = _X.,123,voicemail(100,s) Per the instructions, (100) plays the standard message, (100,b) plays busy (100,u) plays unavailable and (100,s) plays nothing (skip instructions). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, February 21, 2012 10:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Define custom vm-login sound file per VM context? Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? By default the sound file is vm-login.codec. Is there a way to specify which sound file is played per context or some other way to play a different sound file in place of vm-login? I have already replaced the default file and named it the same vm-login.x but still I am only able to play one file, not a different file depending on the VM context I send the caller to. I am sure someone has figured this out so, any shortcut to keep me from frying my brain on this would be appreciated. Thanks! --Todd -- _ -- 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] Define custom vm-login sound file per VM context?
Danny, This seems to be a solution for sending people to leave a voicemail, I need a solution for VoiceMailMain() when people call in to get their messages, change greeting etc. If I use the s option with VoiceMailMain it just skips checking the passcode according to the docs. Thanks for your help though, any similar ideas for VoiceMailMain? I am playing the sound file I need before sending them to VoiceMailMain but then Comedian Mail! plays right after of course. --Todd On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas da...@debsinc.com wrote: I believe this is what you want. Instead of this Exten = _X.,123,Voicemail(100) ** ** Do Exten = _X.,123,playback(your-message) Exten = _X.,123,voicemail(100,s) ** ** Per the instructions, (100) plays the standard message, (100,b) plays busy (100,u) plays unavailable and (100,s) plays nothing (skip instructions).** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier *Sent:* Tuesday, February 21, 2012 10:53 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Define custom vm-login sound file per VM context? ** ** Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? ** ** By default the sound file is vm-login.codec. ** ** Is there a way to specify which sound file is played per context or some other way to play a different sound file in place of vm-login? ** ** I have already replaced the default file and named it the same vm-login.x but still I am only able to play one file, not a different file depending on the VM context I send the caller to. ** ** I am sure someone has figured this out so, any shortcut to keep me from frying my brain on this would be appreciated. ** ** Thanks! ** ** --Todd ** ** -- _ -- 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] Define custom vm-login sound file per VM context?
There was a kludgy solution posted a while back that might work for you. Since Asterisk is multi-lingual you could do this Exten = _X.,123,Set(CHANNEL(language)=fr) Exten = _X.,124,Voicemailmain() This assumes you aren't using fr(French). Just copy /var/lib/asterisk/sounds/en to /var/lib/asterisk/sounds/fr and record your alternate instructions in /var/lib/asterisk/sounds/fr/vm-login.gsm (or whatever codec you are using). Using this work-around you could have as many greetings as you can specify languages for. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, February 21, 2012 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Define custom vm-login sound file per VM context? Danny, This seems to be a solution for sending people to leave a voicemail, I need a solution for VoiceMailMain() when people call in to get their messages, change greeting etc. If I use the s option with VoiceMailMain it just skips checking the passcode according to the docs. Thanks for your help though, any similar ideas for VoiceMailMain? I am playing the sound file I need before sending them to VoiceMailMain but then Comedian Mail! plays right after of course. --Todd On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas da...@debsinc.com wrote: I believe this is what you want. Instead of this Exten = _X.,123,Voicemail(100) Do Exten = _X.,123,playback(your-message) Exten = _X.,123,voicemail(100,s) Per the instructions, (100) plays the standard message, (100,b) plays busy (100,u) plays unavailable and (100,s) plays nothing (skip instructions). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, February 21, 2012 10:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Define custom vm-login sound file per VM context? Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? By default the sound file is vm-login.codec. Is there a way to specify which sound file is played per context or some other way to play a different sound file in place of vm-login? I have already replaced the default file and named it the same vm-login.x but still I am only able to play one file, not a different file depending on the VM context I send the caller to. I am sure someone has figured this out so, any shortcut to keep me from frying my brain on this would be appreciated. Thanks! --Todd -- _ -- 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] Define custom vm-login sound file per VM context?
From: Todd Routhier fonema...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2012 11:30:34 AM
Subject: Re: [asterisk-users] Define custom vm-login sound file per
VM context?
Danny,
This seems to be a solution for sending people to leave a voicemail,
I need a solution for VoiceMailMain() when people call in to get
their messages, change greeting etc.
If I use the s option with VoiceMailMain it just skips checking the
passcode according to the docs.
Thanks for your help though, any similar ideas for VoiceMailMain?
I am playing the sound file I need before sending them to
VoiceMailMain but then Comedian Mail! plays right after of course.
--Todd
The sound files referenced by voicemail.conf are global for all
mailboxes defined in the configuration file, regardless of whether or
not those mailboxes are defined in separate contexts. Hence, whatever
is defined for the 'vm-login' sound will be played for all users.
For this one sound file (and this one sound file only), there is a
mechanism you can use to bypass playing this sound file back. You
can tell VoiceMailMain to skip authentication of the user using the
's' flag, and use VMAuthenticate to authenticate the user yourself.
Note that internally, VoiceMailMain uses VMAuthenticate, so you're
using the exact same mechanism, just from the dialplan. If you pass the
's' flag to VMAuthenticate, it will not play the vm-login sound,
allowing you, if you want, to play a different soundfile.
In general, it would look something like this (please don't expect this
to work verbatim, but it gives you an idea):
exten = 1,1,NoOp()
same = n,Background(Your-sound-file)
same = n,VMAuthenticate(1@default,s)
same = n,GotoIf($[${AUTH_MAILBOX}=1] $[${AUTH_CONTEXT}=default]?auth:failed)
same = n(auth),VoiceMailMain(s)
same = n,Hangup()
same = n(fail),Hangup()
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Define custom vm-login sound file per VM context?
Wow, that makes me wonder if I could do something like: Set(CHANNEL(language)=Cust327) Then create a Language folder named Cust327 and have it just work. Weee... :-) Of course that leads me to think that I could have whole sets of custom sounds for all of Asterisk based on setting this Language bit on the way in. Guess this would all work as long as there is not some requirement in Asterisk that a language setting must be a real country/language code and not something made up. --Todd On Tue, Feb 21, 2012 at 11:37 AM, Danny Nicholas da...@debsinc.com wrote: There was a “kludgy” solution posted a while back that might work for you. Since Asterisk is “multi-lingual” you could do this Exten = _X.,123,Set(CHANNEL(language)=fr) Exten = _X.,124,Voicemailmain() ** ** This assumes you aren’t using fr(French). Just copy /var/lib/asterisk/sounds/en to /var/lib/asterisk/sounds/fr and record your alternate instructions in /var/lib/asterisk/sounds/fr/vm-login.gsm (or whatever codec you are using). Using this work-around you could have as many greetings as you can specify “languages” for. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier *Sent:* Tuesday, February 21, 2012 11:31 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Define custom vm-login sound file per VM context? ** ** Danny, ** ** This seems to be a solution for sending people to leave a voicemail, I need a solution for VoiceMailMain() when people call in to get their messages, change greeting etc. ** ** If I use the s option with VoiceMailMain it just skips checking the passcode according to the docs. ** ** Thanks for your help though, any similar ideas for VoiceMailMain? ** ** I am playing the sound file I need before sending them to VoiceMailMain but then Comedian Mail! plays right after of course. ** ** --Todd ** ** ** ** On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas da...@debsinc.com wrote: I believe this is what you want. Instead of this Exten = _X.,123,Voicemail(100) Do Exten = _X.,123,playback(your-message) Exten = _X.,123,voicemail(100,s) Per the instructions, (100) plays the standard message, (100,b) plays busy (100,u) plays unavailable and (100,s) plays nothing (skip instructions).** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier *Sent:* Tuesday, February 21, 2012 10:53 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Define custom vm-login sound file per VM context? Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? By default the sound file is vm-login.codec. Is there a way to specify which sound file is played per context or some other way to play a different sound file in place of vm-login? I have already replaced the default file and named it the same vm-login.x but still I am only able to play one file, not a different file depending on the VM context I send the caller to. I am sure someone has figured this out so, any shortcut to keep me from frying my brain on this would be appreciated. Thanks! --Todd -- _ -- 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] Define custom vm-login sound file per VM context?
If I recall correctly, it does have to be a real country and a two-letter code, but that still gives you hundreds of variants for this kludge. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, February 21, 2012 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Define custom vm-login sound file per VM context? Wow, that makes me wonder if I could do something like: Set(CHANNEL(language)=Cust327) Then create a Language folder named Cust327 and have it just work. Weee... :-) Of course that leads me to think that I could have whole sets of custom sounds for all of Asterisk based on setting this Language bit on the way in. Guess this would all work as long as there is not some requirement in Asterisk that a language setting must be a real country/language code and not something made up. --Todd On Tue, Feb 21, 2012 at 11:37 AM, Danny Nicholas da...@debsinc.com wrote: There was a kludgy solution posted a while back that might work for you. Since Asterisk is multi-lingual you could do this Exten = _X.,123,Set(CHANNEL(language)=fr) Exten = _X.,124,Voicemailmain() This assumes you aren't using fr(French). Just copy /var/lib/asterisk/sounds/en to /var/lib/asterisk/sounds/fr and record your alternate instructions in /var/lib/asterisk/sounds/fr/vm-login.gsm (or whatever codec you are using). Using this work-around you could have as many greetings as you can specify languages for. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, February 21, 2012 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Define custom vm-login sound file per VM context? Danny, This seems to be a solution for sending people to leave a voicemail, I need a solution for VoiceMailMain() when people call in to get their messages, change greeting etc. If I use the s option with VoiceMailMain it just skips checking the passcode according to the docs. Thanks for your help though, any similar ideas for VoiceMailMain? I am playing the sound file I need before sending them to VoiceMailMain but then Comedian Mail! plays right after of course. --Todd On Tue, Feb 21, 2012 at 10:59 AM, Danny Nicholas da...@debsinc.com wrote: I believe this is what you want. Instead of this Exten = _X.,123,Voicemail(100) Do Exten = _X.,123,playback(your-message) Exten = _X.,123,voicemail(100,s) Per the instructions, (100) plays the standard message, (100,b) plays busy (100,u) plays unavailable and (100,s) plays nothing (skip instructions). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Routhier Sent: Tuesday, February 21, 2012 10:53 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Define custom vm-login sound file per VM context? Is it possible to define a customize the which sound file is played when I send a caller to VoiceMailMain()? By default the sound file is vm-login.codec. Is there a way to specify which sound file is played per context or some other way to play a different sound file in place of vm-login? I have already replaced the default file and named it the same vm-login.x but still I am only able to play one file, not a different file depending on the VM context I send the caller to. I am sure someone has figured this out so, any shortcut to keep me from frying my brain on this would be appreciated. Thanks! --Todd -- _ -- 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] Define custom vm-login sound file per VM context?
OK, this will work and is probably a better solution than the language
idea. Although, the language idea just sounds easier and a little more fun
:-)
Hmm, I think I will try the language solution and see if it works with a
fake country/language code like Cust327 or whatever.
Just wonder if that will break anything else now or with future upgrades.
Thanks for all the help!
--Todd
On Tue, Feb 21, 2012 at 11:47 AM, Matthew Jordan mjor...@digium.com wrote:
From: Todd Routhier fonema...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 21, 2012 11:30:34 AM
Subject: Re: [asterisk-users] Define custom vm-login sound file per
VM context?
Danny,
This seems to be a solution for sending people to leave a voicemail,
I need a solution for VoiceMailMain() when people call in to get
their messages, change greeting etc.
If I use the s option with VoiceMailMain it just skips checking the
passcode according to the docs.
Thanks for your help though, any similar ideas for VoiceMailMain?
I am playing the sound file I need before sending them to
VoiceMailMain but then Comedian Mail! plays right after of course.
--Todd
The sound files referenced by voicemail.conf are global for all
mailboxes defined in the configuration file, regardless of whether or
not those mailboxes are defined in separate contexts. Hence, whatever
is defined for the 'vm-login' sound will be played for all users.
For this one sound file (and this one sound file only), there is a
mechanism you can use to bypass playing this sound file back. You
can tell VoiceMailMain to skip authentication of the user using the
's' flag, and use VMAuthenticate to authenticate the user yourself.
Note that internally, VoiceMailMain uses VMAuthenticate, so you're
using the exact same mechanism, just from the dialplan. If you pass the
's' flag to VMAuthenticate, it will not play the vm-login sound,
allowing you, if you want, to play a different soundfile.
In general, it would look something like this (please don't expect this
to work verbatim, but it gives you an idea):
exten = 1,1,NoOp()
same = n,Background(Your-sound-file)
same = n,VMAuthenticate(1@default,s)
same = n,GotoIf($[${AUTH_MAILBOX}=1]
$[${AUTH_CONTEXT}=default]?auth:failed)
same = n(auth),VoiceMailMain(s)
same = n,Hangup()
same = n(fail),Hangup()
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Define custom vm-login sound file per VM context?
2012-02-21 19:20, Todd Routhier skrev: OK, this will work and is probably a better solution than the language idea. Although, the language idea just sounds easier and a little more fun :-) Hmm, I think I will try the language solution and see if it works with a fake country/language code like Cust327 or whatever. Just wonder if that will break anything else now or with future upgrades. Thanks for all the help! You can also use en_baselevel_customer234 as a language, asterisk will first try to find a soundfile in the en_baselevel_customer234-directory, and if not found in the en_baselevel-directory. After that it will look in the en-dir. Can't find the docs for this right now but this way you don't need to copy all the recordings, and you can stack as many layers as you like.. :-) /Johan -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- 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] Define custom vm-login sound file per VM context?
Wow, that looks like good stuff. On Tue, Feb 21, 2012 at 12:24 PM, Johan Wilfer li...@jttech.se wrote: 2012-02-21 19:20, Todd Routhier skrev: OK, this will work and is probably a better solution than the language idea. Although, the language idea just sounds easier and a little more fun :-) Hmm, I think I will try the language solution and see if it works with a fake country/language code like Cust327 or whatever. Just wonder if that will break anything else now or with future upgrades. Thanks for all the help! You can also use en_baselevel_customer234 as a language, asterisk will first try to find a soundfile in the en_baselevel_customer234-directory, and if not found in the en_baselevel-directory. After that it will look in the en-dir. Can't find the docs for this right now but this way you don't need to copy all the recordings, and you can stack as many layers as you like.. :-) /Johan -- Med vänlig hälsning Johan Wilfer email: jo...@jttech.se JT Tech | Utvecklare webb: http://jttech.se direkt: +46 31 380 91 01 support: +46 31 380 91 00 -- _ -- 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] Set T38 protocol
Hello, I'm trying to send a fax with sendafax aplication and receive the fax with the receiveFax aplication on the same Asterisk Server (1.8..8.2). All work fine but the PBX always use T30 protocol. Is thes a variable or setting to configure Asterisk to send and receive this fax with T38 protocol only? Thank you 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] Streaming musiconhold via mpg123
At my wits end with this, and can't proceed any further so I'm hoping someone has seen this and can assist. I can not get streaming musiconhold to work with Asterisk. My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is CentOS 5.7. When I call the musiconhold class (default for example) I get nothing but silence. I've exhausted my troubleshooting capabilities at this point, I've tried everything I can think of to include: - a newer version of mpg123, I went with the latest version - verified I could play an MP3 file by itself in Asterisk by using the MP3Player application What does not work, is if I use the mpg123 application for musiconhold to play a standalone file or a streaming source. I seem to be missing something and I just can't quite put a finger on it. -- _ -- 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] Streaming musiconhold via mpg123
There is a bug in up to version 1.8.9 with external moh sources and dahdi timers -Original Message- From: Stephen Brown stephen.brow...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 21 Feb 2012 15:34:19 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Streaming musiconhold via mpg123 At my wits end with this, and can't proceed any further so I'm hoping someone has seen this and can assist. I can not get streaming musiconhold to work with Asterisk. My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is CentOS 5.7. When I call the musiconhold class (default for example) I get nothing but silence. I've exhausted my troubleshooting capabilities at this point, I've tried everything I can think of to include: - a newer version of mpg123, I went with the latest version - verified I could play an MP3 file by itself in Asterisk by using the MP3Player application What does not work, is if I use the mpg123 application for musiconhold to play a standalone file or a streaming source. I seem to be missing something and I just can't quite put a finger on it. -- _ -- 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] Streaming musiconhold via mpg123
On Tue, Feb 21, 2012 at 2:34 PM, Stephen Brown stephen.brow...@gmail.comwrote: At my wits end with this, and can't proceed any further so I'm hoping someone has seen this and can assist. I can not get streaming musiconhold to work with Asterisk. My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is CentOS 5.7. When I call the musiconhold class (default for example) I get nothing but silence. I've exhausted my troubleshooting capabilities at this point, I've tried everything I can think of to include: - a newer version of mpg123, I went with the latest version - verified I could play an MP3 file by itself in Asterisk by using the MP3Player application What does not work, is if I use the mpg123 application for musiconhold to play a standalone file or a streaming source. I seem to be missing something and I just can't quite put a finger on it. Share with us your musiconhold.conf configuration please. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- 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] Praking lot issues.
Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant -- _ -- 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] Praking lot issues.
On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] Praking lot issues.
Is it just me, or is doing a blind transfer to a parking lot not such a great idea? If I'm a receptionist, I'm going to want to know the lot number to tell somebody to pick up the call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Tuesday, February 21, 2012 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Praking lot issues. On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] Praking lot issues.
Jason Thank you for the response. It looks as if I am running up against this bug as well. I was also using park more like park and announce and that was giving me issues. I will watch this bug report and make some modifications to my park scenarios. Thanks Bryant From: Jason Parker jpar...@digium.com Sent: Tuesday, February 21, 2012 3:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Praking lot issues. On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] conferenced transfers
On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote: On the snom too Create a conferance and then press the transfer button. That will join the parties and release the receptionist Hmm...You can do that with just hitting the transfer button, or is there more? I'm using a Snom 870 with firmware 8.4.35. I set up the conference, but when I hit transfer, it presents me with a transfer party dialog. It's a bit confusing, because it's not really clear which party is being transferred. Are you using a different phone or firmware? Maybe there's a setting somewhere? -- _ -- 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] Praking lot issues.
Danny I see our point, but we are trying to transfer to a know spot using (BLF) My issue is that it does not appear to work as it keeps looping the call back to the callers extension. I think I may have figured out a way arround that but it will take some more testing. To be sure. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Danny Nicholas da...@debsinc.com Sent: Tuesday, February 21, 2012 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Praking lot issues. Is it just me, or is doing a blind transfer to a parking lot not such a great idea? If I'm a receptionist, I'm going to want to know the lot number to tell somebody to pick up the call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Tuesday, February 21, 2012 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Praking lot issues. On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] Praking lot issues.
How much time are you giving to pick up the lot? I think the default is like 30 seconds. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Tuesday, February 21, 2012 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Praking lot issues. Danny I see our point, but we are trying to transfer to a know spot using (BLF) My issue is that it does not appear to work as it keeps looping the call back to the callers extension. I think I may have figured out a way arround that but it will take some more testing. To be sure. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 _ From: Danny Nicholas da...@debsinc.com Sent: Tuesday, February 21, 2012 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Praking lot issues. Is it just me, or is doing a blind transfer to a parking lot not such a great idea? If I'm a receptionist, I'm going to want to know the lot number to tell somebody to pick up the call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Tuesday, February 21, 2012 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Praking lot issues. On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] Praking lot issues.
Danny We are setting our default to 3 min, but we will allow our users to adjust their setting to what they want for their lot. From our perspective the best time really depends on use case. Say you needed to park a callers for a shop floor. That park time would need to be greater than a general office park time. Thanks Bryant From: Danny Nicholas da...@debsinc.com Sent: Tuesday, February 21, 2012 5:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Praking lot issues. How much time are you giving to pick up the lot? I think the default is like 30 seconds. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Tuesday, February 21, 2012 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Praking lot issues. Danny I see our point, but we are trying to transfer to a know spot using (BLF) My issue is that it does not appear to work as it keeps looping the call back to the callers extension. I think I may have figured out a way arround that but it will take some more testing. To be sure. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: Danny Nicholas da...@debsinc.com Sent: Tuesday, February 21, 2012 4:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Praking lot issues. Is it just me, or is doing a blind transfer to a parking lot not such a great idea? If I'm a receptionist, I'm going to want to know the lot number to tell somebody to pick up the call? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Tuesday, February 21, 2012 3:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Praking lot issues. On 02/21/2012 02:55 PM, Bryant Zimmerman wrote: Ok I now have the basics for dynamic parking working but for some reason when a caller calls in and is parked with a transfer the return call dials the sip peer of the caller and not hte peer of the last party that parked the call. Anyone have any ideas on this? Also when a call is transfered to a parking space. the caller hears the space number. How can I stop that as well? Thanks Bryant See https://issues.asterisk.org/jira/browse/ASTERISK-19322 -- _ -- 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] Streaming musiconhold via mpg123
On 2/21/2012 3:38 PM, isr...@gmail.com wrote: There is a bug in up to version 1.8.9 with external moh sources and dahdi timers Do you have a link to the bug report? I was unable to find anything but it's possible I'm not looking hard enough ;) Share with us your musiconhold.conf configuration please. Here it is... please excuse the mess, it's been a wild ride so my formatting/commenting has been left in-tact: ; ; Music on hold class definitions ; This is using the new 1.2 config file format, and will not work with 1.0 ; based Asterisk systems ; ; #include musiconhold_custom.conf ; #include musiconhold_additional.conf ;[default] ;mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 [test] mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 I setup a simple 2 digit extension to call the test context and my MP3 file nor my stream will play, and here's something else interesting: If use the MP3Player application to play an MP3, mpg123 spawns and plays it. I came to this conclusion by running ps aux | grep mpg while the song was playing. -- _ -- 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] Streaming musiconhold via mpg123
that bug is running since the start of 1.8 and has been fixed in 1.8.9 https://issues.asterisk.org/jira/browse/ASTERISK-17474 i know it says that after the first time asterisks starts it works but thats true only if the moh was loaded before the timing its a long story but the fix is finally in when typing timing test in the cli what timer to get if its dahdi then thats the probably problem On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown stephen.brow...@gmail.comwrote: On 2/21/2012 3:38 PM, isr...@gmail.com wrote: There is a bug in up to version 1.8.9 with external moh sources and dahdi timers Do you have a link to the bug report? I was unable to find anything but it's possible I'm not looking hard enough ;) Share with us your musiconhold.conf configuration please. Here it is... please excuse the mess, it's been a wild ride so my formatting/commenting has been left in-tact: ; ; Music on hold class definitions ; This is using the new 1.2 config file format, and will not work with 1.0 ; based Asterisk systems ; ; #include musiconhold_custom.conf ; #include musiconhold_additional.conf ;[default] ;mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 [test] mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 I setup a simple 2 digit extension to call the test context and my MP3 file nor my stream will play, and here's something else interesting: If use the MP3Player application to play an MP3, mpg123 spawns and plays it. I came to this conclusion by running ps aux | grep mpg while the song was playing. -- _ -- 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] Streaming musiconhold via mpg123
DAHDI it is are there any known workarounds? I use the FreePBX distro and they are a bit behind, so no telling when they will update. On 2/21/2012 6:45 PM, Israel Gottlieb wrote: that bug is running since the start of 1.8 and has been fixed in 1.8.9 https://issues.asterisk.org/jira/browse/ASTERISK-17474 i know it says that after the first time asterisks starts it works but thats true only if the moh was loaded before the timing its a long story but the fix is finally in when typing timing test in the cli what timer to get if its dahdi then thats the probably problem On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown stephen.brow...@gmail.com mailto:stephen.brow...@gmail.com wrote: On 2/21/2012 3:38 PM, isr...@gmail.com mailto:isr...@gmail.com wrote: There is a bug in up to version 1.8.9 with external moh sources and dahdi timers Do you have a link to the bug report? I was unable to find anything but it's possible I'm not looking hard enough ;) Share with us your musiconhold.conf configuration please. Here it is... please excuse the mess, it's been a wild ride so my formatting/commenting has been left in-tact: ; ; Music on hold class definitions ; This is using the new 1.2 config file format, and will not work with 1.0 ; based Asterisk systems ; ; #include musiconhold_custom.conf ; #include musiconhold_additional.conf ;[default] ;mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 [test] mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 I setup a simple 2 digit extension to call the test context and my MP3 file nor my stream will play, and here's something else interesting: If use the MP3Player application to play an MP3, mpg123 spawns and plays it. I came to this conclusion by running ps aux | grep mpg while the song was playing. -- _ -- 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] Streaming musiconhold via mpg123
You could preload the res_moh (don't remember the full name) but that will only help until the next reload which is the next time you'll click the orange bar Or use a different timer which could get you into other problems Maybe some else has a other idea -Original Message- From: Stephen Brown stephen.brow...@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 21 Feb 2012 20:04:29 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Streaming musiconhold via mpg123 -- _ -- 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] Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network?
On 02/17/2012 03:28 AM, Frank Church wrote: Should a Linksys Sipura 2102 be configured with nat=yes even if it is on the local network? I have been having some troubles with a Linksys Sipura 2100 series, which suffers from NO AUDIO after a few calls.. Because it is on the same subnet as Asterisk it is configured with nat=no. When you think of it because the Sipura 2100 is a broadband router, the voice part may be considered as being behind NAT, as are other devices plugged into its yellow socket defintely are. In theory is it likely to be better that way? That was my question exactly. Look for the thread: Should you ever use nat=no? And the answer is almost never. sean -- _ -- 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] Streaming musiconhold via mpg123
On 02/21/2012 05:34 PM, Stephen Brown wrote: application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 Probably unrelated to your issue, but you're going to want to quote that filename. -- _ -- 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] Streaming musiconhold via mpg123
You do need to wait until FreePBX updates the Asterisk. Use yum to install the modules you need. -Vladimir On 2/21/2012 7:04 PM, Stephen Brown wrote: DAHDI it is are there any known workarounds? I use the FreePBX distro and they are a bit behind, so no telling when they will update. On 2/21/2012 6:45 PM, Israel Gottlieb wrote: that bug is running since the start of 1.8 and has been fixed in 1.8.9 https://issues.asterisk.org/jira/browse/ASTERISK-17474 i know it says that after the first time asterisks starts it works but thats true only if the moh was loaded before the timing its a long story but the fix is finally in when typing timing test in the cli what timer to get if its dahdi then thats the probably problem On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown stephen.brow...@gmail.com mailto:stephen.brow...@gmail.com wrote: On 2/21/2012 3:38 PM, isr...@gmail.com mailto:isr...@gmail.com wrote: There is a bug in up to version 1.8.9 with external moh sources and dahdi timers Do you have a link to the bug report? I was unable to find anything but it's possible I'm not looking hard enough ;) Share with us your musiconhold.conf configuration please. Here it is... please excuse the mess, it's been a wild ride so my formatting/commenting has been left in-tact: ; ; Music on hold class definitions ; This is using the new 1.2 config file format, and will not work with 1.0 ; based Asterisk systems ; ; #include musiconhold_custom.conf ; #include musiconhold_additional.conf ;[default] ;mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 [test] mode=custom ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074 application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000 /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3 I setup a simple 2 digit extension to call the test context and my MP3 file nor my stream will play, and here's something else interesting: If use the MP3Player application to play an MP3, mpg123 spawns and plays it. I came to this conclusion by running ps aux | grep mpg while the song was playing. -- _ -- 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
