Re: [asterisk-users] getting error chan_sip.c: Failed to grab lock, trying again..
Hi! I am running asterisk ver 1.2.4 and have faced this error: Try a downgrade to Asterisk 0.7.1 ;- Philipp -- _ -- 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] agi playback to execute say.conf settings
hi, any response ? thanks, Ashik On Mon, Sep 6, 2010 at 12:01 PM, Ashik Ali beaasteriskg...@gmail.comwrote: Hi all, I am able to understand your solutions. Depending upon the india number reading method, I changed number reading setting in say.conf language. For more details visit my blog http://asterisknumbertovoice.blogspot.com/. It is working well with playback(num:123456,say) when I specified it in dialplan. Thanks, Ashik On Thu, Sep 2, 2010 at 7:08 PM, Danny Nicholas da...@debsinc.com wrote: *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *asteriskguru asteriskguru *Subject:* [asterisk-users] agi playback to execute say.conf settings Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. snip but when I write it in agi does not working. Here is agi debug output from asterisk. SIP/6000-000aAGI Rx EXEC playback num:333456,say -- AGI Script Executing Application: (playback) Options: (num:333456,say) SIP/6000-000aAGI Tx 200 result=0 Anybody have any ideas to work it out in agi playback ? Replace playback “num:334456,say” with “say number 334456” Refer to http://www.voip-info.org/wiki/view/say+number -- _ -- 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] IPSec on asterisk
I am not getting anything in debug because call is not reaching us from other end, it is inbound connection over ipsec. Thanks. From: Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com] Sent: 08 September 2010 17:10 To: 'asterisk-users@lists.digium.com' Subject: IPSec on asterisk Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can't receive calls. Can anyone please tell if any extra step is needed. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 Calendar
On Wed, Sep 8, 2010 at 4:24 PM, Adrià Vidal adriavi...@gmail.com wrote: Sorry was my fault , res_calendar was ok, but ical and caldav need other libs (neon,ical...) that were not installed in my system. -- -- Adrià Vidal -- _ -- 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] IPSec on asterisk
I don't know exactly what help you expect to receive in this forum. Asterisk itself has nothing to do with VPNs of any kind, and you should take your questions regarding the setup and configuration of them to the appropriate place. On 09/09/10 18:26, Deepika Nijhawan wrote: I am not getting anything in debug because call is not reaching us from other end, it is inbound connection over ipsec. Thanks. *From:* Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com] *Sent:* 08 September 2010 17:10 *To:* 'asterisk-users@lists.digium.com' *Subject:* IPSec on asterisk Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can’t receive calls. Can anyone please tell if any extra step is needed. Thanks -- _ -- 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] syntax error, unexpected 'token'
Hello list,
getting warning : *syntax error, unexpected 'token'*
dialplan :
exten = pbx,n,Macro(CheckNetworkProblems,${custID})
exten = pbx,n,NoOp(status = ${STATUS})
exten = pbx,n,GoToIf($[${STATUS}=congestion]?backup:nocongestion)
CLI :
[Sep 9 12:27:07] -- Executing [...@cust:15]
NoOp(SIP/test13-002a, status = congestion) in new stack
*[Sep 9 12:27:07] WARNING[5741]: ast_expr2.fl:445 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token',
expecting $end; Input:
congestion=congestion
^*
[Sep 9 12:27:07] WARNING[5741]: ast_expr2.fl:449 ast_yyerror: If you
have questions, please refer to doc/tex/channelvariables.tex.
[Sep 9 12:27:07] -- Executing [...@cust:16]
GotoIf(SIP/test13-002a, ?backup:nocongestion) in new stack
What is then the correct syntax ?!
Jonas.
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Re: [asterisk-users] syntax error, unexpected 'token'
Jonas Kellens wrote:
Hello list,
getting warning : *syntax error, unexpected 'token'*
dialplan :
exten = pbx,n,Macro(CheckNetworkProblems,${custID})
exten = pbx,n,NoOp(status = ${STATUS})
exten = pbx,n,GoToIf($[${STATUS}=congestion]?backup:nocongestion)
CLI :
[Sep 9 12:27:07] -- Executing [...@cust:15]
NoOp(SIP/test13-002a, status = congestion) in new stack
*[Sep 9 12:27:07] WARNING[5741]: ast_expr2.fl:445 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected 'token',
expecting $end; Input:
congestion=congestion
^*
[Sep 9 12:27:07] WARNING[5741]: ast_expr2.fl:449 ast_yyerror: If you
have questions, please refer to doc/tex/channelvariables.tex.
[Sep 9 12:27:07] -- Executing [...@cust:16]
GotoIf(SIP/test13-002a, ?backup:nocongestion) in new stack
What is then the correct syntax ?!
Jonas.
Looks like your ${STATUS} variable contains quotes so you are effectivly
double quoting it in the gotoif command.
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[asterisk-users] How to avoid interruptions with DIGIUM
Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the boot Kernel *pci=routeirq* just to check if this will help? Thanks! -- Salu2 -- _ -- 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] IPSec on asterisk
On Thu, Sep 9, 2010 at 4:26 AM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: I am not getting anything in debug because call is not reaching us from other end, it is inbound connection over ipsec. Asterisk != IPSec Like Rob said, repost your question once you have IPSec working. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] SPA3102 FAX not working
Hi, I have created one SIP extension in Asterisk and configured that extension in SPA 3102. And connected one FAX machine to the SPA3102 and one to Asterisk. The problem is if I try to send FAX from SPA3102 to Asterisk i am not able to send. But if the same if I try to send from Asterisk to SPA3102 it is working. In the SPA3102 following are the options I changed for the FAX settings, Call Waiting Serv:No Three Way Call Serv:No Preferred Codec:G711u FAX Passthru Method:ReINVITE DTMF Tx Mode:Normal FAX Enable T38:yes After all these changes still I am not able to send the fax, while checking in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas from Asterisk to SPA3102 I can able to see some rtp traffic. Your help would be much appreciated On Tue, Sep 7, 2010 at 10:30 PM, Gopalakrishnan A.N sai...@gmail.comwrote: Hi, I have created one SIP extension in Asterisk and configured that extension in SPA 3102. And connected one FAX machine to the SPA3102 and one to Asterisk. The problem is if I try to send FAX from SPA3102 to Asterisk i am not able to send. But if the same if I try to send from Asterisk to SPA3102 it is working. In the SPA3102 following are the options I changed for the FAX settings, Call Waiting Serv:No Three Way Call Serv:No Preferred Codec:G711u FAX Passthru Method:ReINVITE DTMF Tx Mode:Normal FAX Enable T38:yes After all these changes still I am not able to send the fax, while checking in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas from Asterisk to SPA3102 I can able to see some rtp traffic. Your help would be much appreciated -- Thank you with regards, Gopalakrishnan A.N, -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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 to avoid interruptions with DIGIUM
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Thursday, September 09, 2010 6:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to avoid interruptions with DIGIUM Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the boot Kernel pci=routeirq just to check if this will help? Thanks! In my shop we had to recompile the kernel (OpenSuse with TDM410P) because the TDM410P shared IRQ with IDE controller. I would exhaust my Google search before I went that drastic a route. -- _ -- 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] Mirroring or other arangement to secure *
Hi, Please excuse me for addressing this Linux issue on this list, however I hope that some of you have found a solution thats matches the * use and also easy to install without very deep knowledge of Linux. My wish are a program that maintain a mirror copy of the HD. Thank you! Best regards HB -- _ -- 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 to avoid interruptions with DIGIUM
modprobe blacklisting may be of help... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Thu, Sep 9, 2010 at 9:15 AM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias Sent: Thursday, September 09, 2010 6:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to avoid interruptions with DIGIUM Hello Asterisk community, I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm sharing IRQ with some megasas device: 169: 69917985 0 0 0 0 0 0 0 IO-APIC-level megasas, wct4xxp I've been searching here: http://ubuntuforums.org/showthread.php?t=254623 Should i try in the pass this parameter in the boot Kernel pci=routeirq just to check if this will help? Thanks! In my shop we had to recompile the kernel (OpenSuse with TDM410P) because the TDM410P shared IRQ with IDE controller. I would exhaust my Google search before I went that drastic a route… -- _ -- 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] Mirroring or other arangement to secure *
HB wrote: Please excuse me for addressing this Linux issue on this list, however I hope that some of you have found a solution thats matches the * use and also easy to install without very deep knowledge of Linux. My wish are a program that maintain a mirror copy of the HD. http://en.wikipedia.org/wiki/Standard_RAID_levels#RAID_1 Be warned that this is not a replacement for regular backups, because all writes (including accidentally deleting your Asterisk configuration) are done to each disk in the array. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] agi playback to execute say.conf settings
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali
Sent: Thursday, September 09, 2010 2:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi playback to execute say.conf settings
hi,
any response ?
thanks,
Ashik
snip
exec playback in AGI expects to find a file or set of files in
/var/lib/asterisk/sounds. Here is the vital snippet from the PERL AGI I
whipped up to test your query
print EXEC PLAYBACK beep \\\n;
#print EXEC PLAYBACK 'num:334456,say' \\\n; this line will fail with
-- AGI Script Executing Application: (PLAYBACK) Options:
('num:334456,say')
[Sep 9 08:33:16] WARNING[8569]: file.c:664 ast_openstream_full: File
'num:334456,say' does not exist in any format
[Sep 9 08:33:16] WARNING[8569]: file.c:991 ast_streamfile: Unable to open
'num:334456,say' (format 0x4 (ulaw)): No such file or directory
[Sep 9 08:33:16] WARNING[8569]: app_playback.c:440 playback_exec:
ast_streamfile failed on SIP/134-000a for 'num:334456,say'
print SAY NUMBER 334456 \\\n; This does the same thing in AGI as
playback(num:334456,say)
print EXEC PLAYBACK beep \\\n;
my $result = STDIN;
--
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[asterisk-users] Set channel variable from within other channel
Hello list, is it possible to set a variable (channel variable) from within another channel ?! I'm currently working with 2 channels that I bridge afterwards. It would be good to set a variable in one channel when something occurs in the other channel. If some variable is not set in channel 1, then this means something for channel 2. But from within channel 2 I can not see the variables that are set in channel 1. The suggestion of using global variables I think will create difficulties with simultaneous calls... Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA3102 FAX not working
Can you clarify 'send FAX from SPA3102 to Asterisk' ? Are you running Fax for Asterisk software? Where do you expect your call to terminate? As I understand it, without FFA or some of the patches floating around, Asterisk is not T.38 gateway aware. Without more information, it's hard to pinpoint the issue, but as a first step, I would *DISABLE* the 'FAX Enable T38' parameter on the SPA3102. This will not help the reliability, but you may see the calls go through properly. A packet capture is always helpful as well. tcpdump on the Asterisk box or Wireshark on a box connected to your switch's mirror/span port will do. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Gopalakrishnan A.N sai...@gmail.com wrote: Hi, I have created one SIP extension in Asterisk and configured that extension in SPA 3102 . And connected one FAX machine to the SPA3102 and one to Asterisk. The problem is if I try to send FAX from SPA3102 to Asterisk i am not able to send. But if the same if I try to send from Asterisk to SPA3102 it is working. In the SPA3102 following are the options I changed for the FAX settings, Call Waiting Serv:No Three Way Call Serv:No Preferred Codec:G711u FAX Passthru Method:ReINVITE DTMF Tx Mode:Normal FAX Enable T38:yes After all these changes still I am not able to send the fax, while checking in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas from Asterisk to SPA3102 I can able to see some rtp traffic. Your help would be much appreciated On Tue, Sep 7, 2010 at 10:30 PM, Gopalakrishnan A.N sai...@gmail.com wrote: Hi, I have created one SIP extension in Asterisk and configured that extension in SPA 3102 . And connected one FAX machine to the SPA3102 and one to Asterisk. The problem is if I try to send FAX from SPA3102 to Asterisk i am not able to send. But if the same if I try to send from Asterisk to SPA3102 it is working. In the SPA3102 following are the options I changed for the FAX settings, Call Waiting Serv:No Three Way Call Serv:No Preferred Codec:G711u FAX Passthru Method:ReINVITE DTMF Tx Mode:Normal FAX Enable T38:yes After all these changes still I am not able to send the fax, while checking in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas from Asterisk to SPA3102 I can able to see some rtp traffic. Your help would be much appreciated -- Thank you with regards, Gopalakrishnan A.N, -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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 to avoid interruptions with DIGIUM
- Andrew Latham lath...@gmail.com wrote: modprobe blacklisting may be of help... The module sharing interrupts with the card is his storage controller (megasas). Blacklisting the storage controller module? That is not a good idea... Try putting the card in another slot. --Tim -- _ -- 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] info about application not available asterisk 1.6.2.11
Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set channel variable from within other channel
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, September 09, 2010 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set channel variable from within other channel
Hello list,
is it possible to set a variable (channel variable) from within another
channel ?!
I'm currently working with 2 channels that I bridge afterwards. It would be
good to set a variable in one channel when something occurs in the other
channel.
If some variable is not set in channel 1, then this means something for
channel 2. But from within channel 2 I can not see the variables that are
set in channel 1.
The suggestion of using global variables I think will create difficulties
with simultaneous calls...
Kind regards,
Jonas.
AFAIK, it is not possible to set a local variable for 1 call from another.
If GLOBAL variables are a concern, why not use the ASTDB to store/retrieve
these values?
exten = 1234,1,Set(DB(passval/${EXTEN}/val1)=1) will create a key
passval/channelname/val with value 1
on further reflection
exten = 1234,1,Set(DB(passval/${EXTEN:0:7}/val1)=1) might create a better
reference key
For a call on SIP/170, line 1 would create passval/SIP-170abcdefg/val1, line
2 would create passval/SIP-170/val1
To see what the channel wrote, you would need to get the bridged channel
value (perhaps core show channels verbose?) and do
Exten = 3456,1,Set(CHAN2=bridged channel)
Exten = 3456,n,Set(TEST2=${DB(passval/${CHAN2}/val1)
Regards,
Danny Nicholas
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Re: [asterisk-users] How to avoid interruptions with DIGIUM
On 09/09/2010 09:01 AM, Tim Nelson wrote: - Andrew Latham lath...@gmail.com wrote: modprobe blacklisting may be of help... The module sharing interrupts with the card is his storage controller (megasas). Blacklisting the storage controller module? That is not a good idea... No, but he may have other devices he isn't actually using that are consuming interrupts, and disabling one or more of those may free up interrupts to be utilized by the storage controller or the TDM interface card. On some systems, though, the possible interrupt choices are restricted per PCI/PCI Express slot, though, so yeah... moving the card to another slot may also be required. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] SPA3102 FAX not working
Hi Tim, Thanks for your reply. I am not using any fax software. I just created two extensions in trixbox (Asterisk 1.4), one extension I configured in one SPA3102 here FAX machine is connected in the phone port of SPA3102. As the same the other extension is configured in another SPA3102 and FAX machine is connected in the phone port of SPA3102. I am sending FAX from one extension to another extension. I am not able to send. This is how my workflow is designed. On Thu, Sep 9, 2010 at 7:29 PM, Tim Nelson tnel...@rockbochs.com wrote: Can you clarify 'send FAX from SPA3102 to Asterisk' ? Are you running Fax for Asterisk software? Where do you expect your call to terminate? As I understand it, without FFA or some of the patches floating around, Asterisk is not T.38 gateway aware. Without more information, it's hard to pinpoint the issue, but as a first step, I would *DISABLE* the 'FAX Enable T38' parameter on the SPA3102. This will not help the reliability, but you may see the calls go through properly. A packet capture is always helpful as well. tcpdump on the Asterisk box or Wireshark on a box connected to your switch's mirror/span port will do. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Gopalakrishnan A.N sai...@gmail.com wrote: Hi, I have created one SIP extension in Asterisk and configured that extension in SPA 3102. And connected one FAX machine to the SPA3102 and one to Asterisk. The problem is if I try to send FAX from SPA3102 to Asterisk i am not able to send. But if the same if I try to send from Asterisk to SPA3102 it is working. In the SPA3102 following are the options I changed for the FAX settings, Call Waiting Serv:No Three Way Call Serv:No Preferred Codec:G711u FAX Passthru Method:ReINVITE DTMF Tx Mode:Normal FAX Enable T38:yes After all these changes still I am not able to send the fax, while checking in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas from Asterisk to SPA3102 I can able to see some rtp traffic. Your help would be much appreciated On Tue, Sep 7, 2010 at 10:30 PM, Gopalakrishnan A.N sai...@gmail.comwrote: Hi, I have created one SIP extension in Asterisk and configured that extension in SPA 3102. And connected one FAX machine to the SPA3102 and one to Asterisk. The problem is if I try to send FAX from SPA3102 to Asterisk i am not able to send. But if the same if I try to send from Asterisk to SPA3102 it is working. In the SPA3102 following are the options I changed for the FAX settings, Call Waiting Serv:No Three Way Call Serv:No Preferred Codec:G711u FAX Passthru Method:ReINVITE DTMF Tx Mode:Normal FAX Enable T38:yes After all these changes still I am not able to send the fax, while checking in the rtp debug from SPA3102 to Asterisk the sequential no. 0 but whereas from Asterisk to SPA3102 I can able to see some rtp traffic. Your help would be much appreciated -- Thank you with regards, Gopalakrishnan A.N, -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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 -- Thank you with regards, Gopalakrishnan A.N, -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Set channel variable from within other channel
On 09/09/2010 04:12 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, September 09, 2010 8:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set channel variable from within other channel
Hello list,
is it possible to set a variable (channel variable) from within
another channel ?!
I'm currently working with 2 channels that I bridge afterwards. It
would be good to set a variable in one channel when something occurs
in the other channel.
If some variable is not set in channel 1, then this means something
for channel 2. But from within channel 2 I can not see the variables
that are set in channel 1.
The suggestion of using global variables I think will create
difficulties with simultaneous calls...
Kind regards,
Jonas.
AFAIK, it is not possible to set a local variable for 1 call from
another. If GLOBAL variables are a concern, why not use the ASTDB to
store/retrieve these values?
exten = 1234,1,Set(DB(passval/${EXTEN}/val1)=1) will create a key
passval/channelname/val with value 1
on further reflection
exten = 1234,1,Set(DB(passval/${EXTEN:0:7}/val1)=1) might create a
better reference key
For a call on SIP/170, line 1 would create
passval/SIP-170abcdefg/val1, line 2 would create passval/SIP-170/val1
To see what the channel wrote, you would need to get the bridged
channel value (perhaps core show channels verbose?) and do
Exten = 3456,1,Set(CHAN2=bridged channel)
Exten = 3456,n,Set(TEST2=${DB(passval/${CHAN2}/val1)
Regards,
Danny Nicholas
Danny,
the wiki mentions :
Set(DB(family/key)=${foo})
What is this 'passval' you are talking about ?!
Kind regards,
Jonas.
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Re: [asterisk-users] Set channel variable from within other channel
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, September 09, 2010 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set channel variable from within other channel
On 09/09/2010 04:12 PM, Danny Nicholas wrote:
_
size=2 width=100% align=center
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, September 09, 2010 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set channel variable from within other channel
Hello list,
is it possible to set a variable (channel variable) from within another
channel ?!
I'm currently working with 2 channels that I bridge afterwards. It would be
good to set a variable in one channel when something occurs in the other
channel.
If some variable is not set in channel 1, then this means something for
channel 2. But from within channel 2 I can not see the variables that are
set in channel 1.
The suggestion of using global variables I think will create difficulties
with simultaneous calls...
Kind regards,
Jonas.
AFAIK, it is not possible to set a local variable for 1 call from another.
If GLOBAL variables are a concern, why not use the ASTDB to store/retrieve
these values?
exten = 1234,1,Set(DB(passval/${EXTEN}/val1)=1) will create a key
passval/channelname/val with value 1
on further reflection
exten = 1234,1,Set(DB(passval/${EXTEN:0:7}/val1)=1) might create a better
reference key
For a call on SIP/170, line 1 would create passval/SIP-170abcdefg/val1, line
2 would create passval/SIP-170/val1
To see what the channel wrote, you would need to get the bridged channel
value (perhaps core show channels verbose?) and do
Exten = 3456,1,Set(CHAN2=bridged channel)
Exten = 3456,n,Set(TEST2=${DB(passval/${CHAN2}/val1)
Regards,
Danny Nicholas
Danny,
the wiki mentions :
Set(DB(family/key)=${foo})
What is this 'passval' you are talking about ?!
Kind regards,
Jonas.
Just being a bit over(under?) verbose) . Since I'm using ASTDB to pass
values between channels, my thought was that the family would be passval
and the key would be ${EXTEN}/val1 (so you could pass multiple values if
desired). In a two-channel environment, it wouldn't be that relevant, but
since I assume you would want to be able to monitor X number of bridged
calls, the 3 value family/key combination made the most sense to me.
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Re: [asterisk-users] SPA3102 FAX not working
Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote: I am sending FAX from one extension to another extension. I am not able to send. Preferred Codec:G711u You forget to mentoin where do you live? In some countries the G711a codec and in onther countries the G711u codec useable. -- Best regards, Gergomailto:csi...@gmail.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] How to avoid interruptions with DIGIUM
Thanks Kevin, But today i saw a Kernel Panic into my server, for no any apparent reasondoes this parameter could help: pci=routeirq By the way, we are using DELL servers, i've also used Sangoma, and always the same problem Thanks! 2010/9/9 Kevin P. Fleming kpflem...@digium.com On 09/09/2010 09:01 AM, Tim Nelson wrote: - Andrew Latham lath...@gmail.com wrote: modprobe blacklisting may be of help... The module sharing interrupts with the card is his storage controller (megasas). Blacklisting the storage controller module? That is not a good idea... No, but he may have other devices he isn't actually using that are consuming interrupts, and disabling one or more of those may free up interrupts to be utilized by the storage controller or the TDM interface card. On some systems, though, the possible interrupt choices are restricted per PCI/PCI Express slot, though, so yeah... moving the card to another slot may also be required. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- Salu2 -- _ -- 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] vegastream 50 BRI-s latest firmware ?
Hi All, sorry for the off topic. I own 3 vegastream 50 BRI-s 4 ISDN ports gateways with firmware version 6 and I need some advanced parameters available only from firmware version 7. I am sure that I need those parameters because changing the vega gateway with a 20$ cologne pci card in an Asterisk box results in a correct call setup. The vendor refuses to provide the firmware because the product is discontinued, also begging at the phone gave the same result: they kindly suggest to replace the units with newer ones, at 900USD each. I purchased them in the 2004. Question: maybe someone happens to have the latest firmware file and may share it with me ? Thanks and regards, Mike -- _ -- 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] SPA3102 FAX not working
I am from India and I hope I have to use G711u...If I am not wrong On Thu, Sep 9, 2010 at 8:36 PM, Gergo Csibra csi...@gmail.com wrote: Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote: I am sending FAX from one extension to another extension. I am not able to send. Preferred Codec:G711u You forget to mentoin where do you live? In some countries the G711a codec and in onther countries the G711u codec useable. -- Best regards, Gergomailto:csi...@gmail.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 -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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] Issues with in-call DTMF using Broadvox and Level 3
The issue we are having is that in-call RFC2833 DTMF digits are being dropped with Broadvox and Level 3. This is happening with Grandstream GXP and Snom phones. We did some testing with the vendors and here is one of the responses we got back. Is there any way to force asterisk to modify the DTMF so that these phones will work with the carriers at issue. This is a big compatibility issue with SONUS. Hello, We have reviewed both captures, one of a good call with proper DTMF passing and one with DTMF failing to send properly. The issue still remains the same. On the good call, the time between the last RTP packet and the first DTMF event was less than 2ms. For the bad call, the time was over 200ms. Also, to clarify, the 100ms requirement is not per RFC but a standard that our switch vendor has put in front of us in order to guarantee proper DTMF passing. We have had them troubleshoot this in the past, but unfortunately it is something they cannot rectify on their end. The next course of action now is to see what can be done to work around this issue. Here are the following options: 1. Get the time between the packets down to ~100ms or lower. 2. Send DTMF via SIP INFO 3. Send DTMF via Inband. Please advise our NOC how you would like to proceed. Thank you, -- _ -- 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] info about application not available asterisk 1.6.2.11
On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens jonas.kell...@telenet.be wrote: asterisk*CLI core show application Dial did you have libxml-doc installed when you build asterisk? *CLI module load app_dial.so -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] info about application not available asterisk 1.6.2.11
On Thu, Sep 9, 2010 at 11:37 AM, Paul Belanger paul.belan...@polybeacon.com wrote: did you have libxml-doc installed when you build asterisk? s/-doc/-dev -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] 3Com 3102 Phones
Does anyone have a packet capture of a 3Com 3102 phone registering with an NBX that I could take a look at ? What is the expected traffic flow, all I get is the 0x8836 initial packet from the phone but have no NBX to validate Please still seeking help No one using 3com with asterisk ?? Thanks On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote: Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they are useless without a NBX or NCP ? Any specs on how to configure linux to act like one? Thanks in advance -- _ -- 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] info about application not available asterisk 1.6.2.11
On 09/09/2010 05:37 PM, Paul Belanger wrote: On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellensjonas.kell...@telenet.be wrote: asterisk*CLI core show application Dial did you have libxml-doc installed when you build asterisk? *CLI module load app_dial.so Indeed I did not had libxml-doc installed and so I build asterisk 1.6.2.11 without it. I did not know the consequence when installing... Can I install libxml-doc now without having to rebuild asterisk ?! Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3Com 3102 Phones
I hadn't heard about them until your mailing list post. On Thu, Sep 9, 2010 at 8:48 AM, Barry Fawthrop ba...@isscp.com wrote: Does anyone have a packet capture of a 3Com 3102 phone registering with an NBX that I could take a look at ? What is the expected traffic flow, all I get is the 0x8836 initial packet from the phone but have no NBX to validate Please still seeking help No one using 3com with asterisk ?? Thanks On Wed, 2010-09-01 at 09:09 -0400, Barry Fawthrop wrote: Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they are useless without a NBX or NCP ? Any specs on how to configure linux to act like one? Thanks in advance -- _ -- 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] info about application not available asterisk 1.6.2.11
On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Can I install libxml-doc now without having to rebuild asterisk ?! No, install libxml-dev then rerun ./configure, make install -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] info about application not available asterisk1.6.2.11
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Subject: Re: [asterisk-users] info about application not available asterisk1.6.2.11 On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote: Can I install libxml-doc now without having to rebuild asterisk ?! No, install libxml-dev then rerun ./configure, make install -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com The reason for this is that the configuration in place is not aware of libxml, so just re-making Asterisk without configuring will result in a continuation of the existing problem. -- _ -- 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] VoIP friendly Internet providers in Dallas and Philadelphia
Hi Everyone, My experience is only with the Canadian providers. What options/providers are there in Dallas and Philadelphia other than Verizon when it comes to internet? Something in the order of at least 10mbps down and up - I understand that and higher bandwidths are easily available in USA due to vast fiber networks? The connection will be replacing a T1 and will be support VPN connections to connect office for VoIP. Thanks -- _ -- 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] Queue/Dial Recording - Capture answering channel name.
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs on the calling channel, and I
need the called channel - Is this accessible?
Thanks,
Steve
Where ever the MixMonitor recording is done add in the ${CHANNEL}
variable to the filename parameter. Or even add in the line below to the
context that contains Dial(QueueName).
For example:
exten =
s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49)
Hope this helps.
Regards,
Antonio
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Re: [asterisk-users] Call Center: scripting for call routing, reporting, login and logout, CTI
bilal ghayyad wrote: Hi All; I would like to use Asterisk for a call center, but really does not know if Asterisk support the following in a good way: 1) Ability to do an inteligent routing, so to route the call to the proper skill group based on the caller information? 2) If I can create skill groups and then the agent will login to this skill group. 3) What about reporting to check the call center performance? How can I get it? 4) To have integration with the CRM, how to be done? Is it using CTI or how? 5) Is it possible that agent to login and logout and be ready and not ready? Appreciate your kindly advise and help. Regards Bilal Yes to all 5. Asterisk can do all the above. I would only use ViciDial if you have a very large call center with many campaigns running, it does add another layer of complexity. p style=margin: 0; padding: 0; border-collapse: collapse; font-family: Tahoma, Arial, Sans-Serif; font-size: 10px; color: #33; ---DISCLAIMERbr / The information contained in this message is private and confidential and intended only for the recipient named above. If you are not the intended recipient you are notified that any communication, circulation or copying of the information contained in this message is strictly prohibited. If you have received this message in error please notify us immediately by telephone in order that we are made aware of this fact and the message can be returned to us at our address as indicated above. Activity and use of the Sheffield City Taxis e-mail service is monitored to secure its effective operation and for other lawful business purposes. Sheffield City Taxis Ltd. Registered Office: 912 City Road, Sheffield, S2 1GQ. Registered in England no: 4674148. Sheffield City Taxis Limited uses regularly updated anti-virus software in an attempt to reduce the possibility of infection. However we do not guarantee that any attachments to this e-mail are virus free./p -- _ -- 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] Queue/Dial Recording - Capture answering channel name.
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs on the calling channel, and I
need the called channel - Is this accessible?
Thanks,
Steve
Where ever the MixMonitor recording is done add in the ${CHANNEL}
variable to the filename parameter. Or even add in the line below to the
context that contains Dial(QueueName).
For example:
exten =
s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49)
I was under the impression from the documentation that the ${CHANNEL}
variable held the _Calling_ channel, I need the _Called_ channel (the
channel that eventually picks up the call)
Am I mistaken? Perhaps I should give it a try :)
Thanks,
Steve
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[asterisk-users] Archive of security advisories?
Is there an archive of security advisories for Asterisk? We recently upgraded a customer from 1.2 to 1.4 and now they are asking for documentation of all security and bug related fixes. I know the advisories get published on this list but is there an easier way to find them than trying to search the list. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Archive of security advisories?
On Thu, Sep 9, 2010 at 10:25 AM, Carlos Chavez cur...@telecomabmex.comwrote: Is there an archive of security advisories for Asterisk? We recently upgraded a customer from 1.2 to 1.4 and now they are asking for documentation of all security and bug related fixes. I know the advisories get published on this list but is there an easier way to find them than trying to search the list. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 The archive is here: http://downloads.asterisk.org/pub/security/ http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35 and search for ASA- example entry: 2007-08-07 18:25 + [r78375] Jason Parker jpar...@digium.com * channels/chan_skinny.c: Properly check the capabilities count to avoid a segfault. (ASA-2007-019) http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ has change logs from 1.2 Looks like 1.4 was started before Asterisk 1.2.13, hopefully they're not asking for a refactored changelog from asterisk 1.2.19 to 1.4.32 ;) -- _ -- 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] Archive of security advisories?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Subject: [asterisk-users] Archive of security advisories? Is there an archive of security advisories for Asterisk? We recently upgraded a customer from 1.2 to 1.4 and now they are asking for documentation of all security and bug related fixes. I know the advisories get published on this list but is there an easier way to find them than trying to search the list. IMO you should be able to get this from the CHANGELOG with the version you install. -- _ -- 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] Archive of security advisories?
On Thu, Sep 09, 2010 at 12:25:03PM -0500, Carlos Chavez wrote: Is there an archive of security advisories for Asterisk? We recently upgraded a customer from 1.2 to 1.4 and now they are asking for documentation of all security and bug related fixes. I know the advisories get published on this list but is there an easier way to find them than trying to search the list. Recent ones: http://www.asterisk.org/security Back to 2007: http://downloads.asterisk.org/pub/security/ -- Barry -- _ -- 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] VoIP friendly Internet providers in Dallas and Philadelphia
As far as Dallas, it completely depends on where you are. The only provider that blankets an area with fiber is Verizon and that is really only 2-3 cities around Dallas and it is usually residential, not business. They aren't in Dallas itself. Time Warner and Cogent have a lot of coverage in bigger buildings, so you might check them. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce Sent: Thursday, September 09, 2010 11:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] VoIP friendly Internet providers in Dallas and Philadelphia Hi Everyone, My experience is only with the Canadian providers. What options/providers are there in Dallas and Philadelphia other than Verizon when it comes to internet? Something in the order of at least 10mbps down and up - I understand that and higher bandwidths are easily available in USA due to vast fiber networks? The connection will be replacing a T1 and will be support VPN connections to connect office for VoIP. Thanks -- _ -- 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] Mirroring or other arangement to secure *
On Thu, 2010-09-09 at 15:29 +0200, hbk wrote: My wish are a program that maintain a mirror copy of the HD. http://www.drbd.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
[asterisk-users] Cisco 7975g running 8.3.4
Have a Cisco 7975g running SIP firmware version 8.3.4. Many things are broken with Asterisk. 1) BLF doesn't work 2) MWI doesn't work 3) Sometimes the calls get stuck on the display 4) Sometimes MOH works 5) Headset jack doesn't work Can anyone recommend a version of the SIP firmware for the Cisco 7975g that they know works well with Asterisk? -- _ -- 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] Archive of security advisories?
On Thursday 09 September 2010 12:46:10 Kyle Kienapfel wrote: On Thu, Sep 9, 2010 at 10:25 AM, Carlos Chavez cur...@telecomabmex.comwrote: Is there an archive of security advisories for Asterisk? We recently upgraded a customer from 1.2 to 1.4 and now they are asking for documentation of all security and bug related fixes. I know the advisories get published on this list but is there an easier way to find them than trying to search the list. The archive is here: http://downloads.asterisk.org/pub/security/ http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.35 and search for ASA- example entry: 2007-08-07 18:25 + [r78375] Jason Parker jpar...@digium.com * channels/chan_skinny.c: Properly check the capabilities count to avoid a segfault. (ASA-2007-019) http://downloads.asterisk.org/pub/telephony/asterisk/old-releases/ has change logs from 1.2 Looks like 1.4 was started before Asterisk 1.2.13, hopefully they're not asking for a refactored changelog from asterisk 1.2.19 to 1.4.32 ;) Shortly after we used the ASA moniker, we changed to using AST to avoid a conflict with another vendor's security advisories, which used the ASA notation prior. We additionally backported all existing advisories which used the ASA notation to AST, so all advisories should be found with the AST notation. You're right about the changelogs, though, so we'll look at fixing those at the download site to ensure that it's consistent. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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
[asterisk-users] Curious what 'early media' is in terms of Answer()
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Answer Can someone clarify what early media is? I noticed that NOT answering a call before dumping them into a queue that has music on hold will not set up a leg to push music back over the calling SIP channel. Tossing an Answer command into the dialplan just before moving to the queue alleviates this (in either situation the queue still works when someone picks up the queued call). This is more of a trivial question, but I'm interested in knowing where (if anywhere, or if it's played at all) the music on hold is going when there is no Answer command. Incidentally, in 1.6.1.x the Answer appeared to be explicit after dumping a call into a queue, which is how I came across this issue after upgrading to 1.6.2.11. hose -- _ -- 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] DAHDI fxstest?
Greetings all- During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? Thanks! --Tim -- _ -- 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] Curious what 'early media' is in terms of Answer()
On Thu, Sep 9, 2010 at 4:38 PM, Hose hose+aster...@bluemaggottowel.com wrote: Can someone clarify what early media is? Basically playing audio to the channel before actually answering the channel (IE: Answer()). You usually use Progress() at the start of your dial plan to send 183 Session Progress SIP Message. Usually used to play audio to the line, by passing toll charges as a result of using Answer(). -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] DAHDI fxstest?
On Thu, Sep 9, 2010 at 4:40 PM, Tim Nelson tnel...@rockbochs.com wrote: Can anyone tell me how to build fxstest? No, but if you output the error message we can help point you in the right direction. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] DAHDI fxstest?
On 9/9/10 1:40 PM, Tim Nelson wrote: During some recent testing and debugging, I wanted to use the 'fxstest' application. However, I found it hasn't been built when doing the standard 'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0... Can anyone tell me how to build fxstest? cd to tools directory then enter the command make menuselect select fxstest, save exit, then execute make. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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 to avoid interruptions with DIGIUM
On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Kevin, But today i saw a Kernel Panic into my server, for no any apparent reasondoes this parameter could help: pci=routeirq By the way, we are using DELL servers, i've also used Sangoma, and always the same problem Thanks! I'd like to know which problem you had with the Sangoma card as there are no shared interrupt issues we know of. There used to be a problem with some Dell servers though, but that was already fixed some weeks ago. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.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
