Re: [asterisk-users] blacklist caller ID
On Thursday, March 14, 2013, Joseph wrote:
Can someone refresh my memory how to backlist caller ID in
asterisk 1.8?
I had it working in ver. 1.4 but in 1.8 it changed.
I'm still using 1.4. In that I add a number to the blacklist with
CLI database put blacklist 0123456789 1
That is to add the number to the blacklist 'group' and give it a value
of 1.
Then is extensions.conf:
exten = main,n,GotoIf($[${BLACKLIST()}]?banned,1)
;(...)
; Blacklisted numbers
exten = banned,1,Playback(silence/2im-sorry)
exten = banned,n,Playback(cannot-complete-as-dialed)
exten = banned,n,Playback(privacy-you-are-blacklisted)
exten = banned,n,Playback(goodbye)
exten = banned,n,Hangup()
Hopefully, the same thing works in 1.8 and 10 (I'd be interested to
know because I intend to upgrade my hardware soon and will probably
take the opportunity to update * to a later version).
HTH,
--
Geoff
--
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Re: [asterisk-users] Recording with MixMonitor and AGI
Hi, The idea was to record an ongoing call by three party bridging on the mobile phone. Well my problem was to halt execution of the Dialplan so the server would not hang up the call. And I don´t want the server to say anything during the call. Now I solved this case as well by using Answer and then Record in the dialplan . So I´m not recording with MixMonitor. But just out of curiosity. How did you mean using hold (in answer/hold). Is that MusicOnHold? For me I can´t use that since I don´t want to make any noise. Is there another way? exten = 111,1,Answer() exten = 111,n,? I have tried using Wait with a long duration but have not succeeded to make it work as I want. I am using asterisk-java and originate calls to local channels. Regards, Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.semailto:henrik.westerb...@ain.se Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again... hu.. I don´t understand.. what do you mean with merging to a mobile phone? do you want do bridge the calls (three partys) or do you want to play the just recorded file from your server-initiated call into a another running call? what is by hand? the more explicit you are, the more helpful will be the answer. you ask but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up of course you can...you could e.g.: call into a queue call into a meetme room call with the help of a local channel into a context where you do nothing but answer / hold but as i said i did not quite catch what your objective really is... i just dont understand your scenario or cant imagine its sense. if you are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1. so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? Yes this is correct, and I prefer to do the transferring of the file to another server with my existing AGI. My AGIs are written in java. Today I the upload is done over http. Today I schedule the upload in the AGI script a couple of seconds after the channel is hang up. But the two lines might not be hung up at the same time. Your suggestion of always fixing the file is wise, it now seems to work fine after having been processed with sox. So now I think that this case 1 is ok for me :-) 2. and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? The idea is to perform a probe call with the only task of recording what the other party says. It will be merged by hand on a mobile phone to an ongoing call with another party. This could be done by calling out and letting AGI execute a RECORD FILE but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up Then I could easily use the MixMonitor and write the whole conversation in the dialplan with uploading similar to the first case. Any suggestions? Regards, Henrik Från: Yves A. yves...@gmx.demailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.commailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? let me know, if i understood you right, the solution is not so hard to implement. In what language do you preferrably write your AGIs? (although there is no absolute need for using an agi... you can all write down in your dialplan...) is there a special protocol requirement for saving/transferring the recorded voicefile (e.g. ftps)? One obstacle is, that the recorded file is not fully
Re: [asterisk-users] Recording with MixMonitor and AGI
hi, the music heard by MoH is configurable... so if you want silence... But hold could e.g. also be done by transferring a caller into a dynamic meetme room... yves Am 14.03.2013 08:43, schrieb Henrik Westerberg: Hi, The idea was to record an ongoing call by three party bridging on the mobile phone. Well my problem was to halt execution of the Dialplan so the server would not hang up the call. And I don´t want the server to say anything during the call. Now I solved this case as well by using Answer and then Record in the dialplan . So I´m not recording with MixMonitor. But just out of curiosity. How did you mean using hold (in answer/hold). Is that MusicOnHold? For me I can´t use that since I don´t want to make any noise. Is there another way? exten = 111,1,Answer() exten = 111,n,? I have tried using Wait with a long duration but have not succeeded to make it work as I want. I am using asterisk-java and originate calls to local channels. Regards, Henrik Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de Datum: söndag 10 mars 2013 11:42 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com, Henrik Westerberg henrik.westerb...@ain.se mailto:henrik.westerb...@ain.se Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI Hi, so if your are ok with the way you solved part 1... alright, lets go to part 2.. but again... hu.. I don´t understand.. what do you mean with merging to a mobile phone? do you want do bridge the calls (three partys) or do you want to play the just recorded file from your server-initiated call into a another running call? what is by hand? the more explicit you are, the more helpful will be the answer. you ask but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up of course you can...you could e.g.: call into a queue call into a meetme room call with the help of a local channel into a context where you do nothing but answer / hold but as i said i did not quite catch what your objective really is... i just dont understand your scenario or cant imagine its sense. if you are a java programmer, i think your using the asterisk-java lib from s. reuter.. if so, you have any freedom, you could also use ami connection to listen to events to start and stop recordings and so on. regards, yves Am 09.03.2013 21:32, schrieb Henrik Westerberg: Hi, Thanks for your answer! 1. so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? Yes this is correct, and I prefer to do the transferring of the file to another server with my existing AGI. My AGIs are written in java. Today I the upload is done over http. Today I schedule the upload in the AGI script a couple of seconds after the channel is hang up. But the two lines might not be hung up at the same time. Your suggestion of always fixing the file is wise, it now seems to work fine after having been processed with sox. So now I think that this case 1 is ok for me :-) 2. and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? The idea is to perform a probe call with the only task of recording what the other party says. It will be merged by hand on a mobile phone to an ongoing call with another party. This could be done by calling out and letting AGI execute a RECORD FILE but if there is a way to just dial out and then let the server side of the call Keep the channel up but do nothing forever until the call is hang up Then I could easily use the MixMonitor and write the whole conversation in the dialplan with uploading similar to the first case. Any suggestions? Regards, Henrik Från: Yves A. yves...@gmx.de mailto:yves...@gmx.de Svara till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Datum: torsdag 7 mars 2013 20:10 Till: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Ämne: Re: [asterisk-users] Recording with MixMonitor and AGI hi, hard to understand, what your objective is... at least for me ;-) so you want to establish a call (triggered by ami) between two partys, record the conversation and save the file to a(nother) server (afterwards), right? and another task is to establish (also ami triggered) a call to a mobile and play, lets say a voicefile. this conversation should also be recorded and saved on a(nother) server (afterwards), right? let me know, if i understood you right, the solution is not so hard to implement. In what language do you preferrably write your AGIs? (although
Re: [asterisk-users] Sending SMS from asterisk
chan_datacard was discontinnued two years ago, chan_dongle is the current dongle driver for asterisk. chan_mobile uses bluetooth mobile phones as FXO devices. to send SMS chan_dongle should be used. i.e: asterisk -rx [ENTER] dongle sms dongle0 0415340999 hello world this command will send and SMS to 0415340999 by dongle0. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/14/2013 9:15 AM, Asghar Mohammad wrote: HI bilal, I don't think DAHDI can send SMS you have 2 options chan_mobile or chan_datacard ex chan_dongle chan_datacard i have not tested but with some mobile phones you can send sms i have tested also with some made in china unbranded phone that are capable to send and receive sms but not good for call termination, they send answer on connect. not all BT dongles are compatible you should go to trail and error for finding combination of dongle and phone. PS: yesterday tested asterisk 11 with chan_mobile and worked without any modification. On Wed, Mar 13, 2013 at 10:29 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi Asghar; I was looking to use chan_mobile for sending SMS, is it possible? Or it is only for calls? By the way, if I have GSM adaptor that convert from SIM card to FXS port, then who I need chan_mobile? I can use DAHDI. So when to use chan_mobile? Regards Bilal - HI Bilal, i am using chan_mobile for call termination, you can use it but you need to tweak chan_mobile.c it is broken from a long time. let me know if you want give it a try. On Mon, Mar 11, 2013 at 6:22 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: - What are the elements of this solution? Is it only: 3G dongles and chan_dongle only? Or there are something else? Bash and perl programing, asterisk and chan_dongle. * Bash and perl programing to do what? It is going to use AMI instead of sending the messages from the commands given in the extensions.conf? Why to use chan_dongle and not chan_mobile? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] PRI Called Party Number Info
Hi, I need to get type of called number (TON), which is displayed in pri debug messages: Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ] Does anyone know how to do it? According to documentation it is only possible for calling number. But I need to make decision in dialplan upon the value of type of called number. BTW, I made a little research on source code and could not find anything related to my question. Perhaps, it's not implemented. Best regards, Grigoriy -- С уважением, Григорий Пузанкин -- _ -- 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] PRI Called Party Number Info
2013/3/14 Puzankin Grigoriy gpuzan...@gmail.com: Hi, I need to get type of called number (TON), which is displayed in pri debug messages: Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ] Does anyone know how to do it? According to documentation it is only possible for calling number. But I need to make decision in dialplan upon the value of type of called number. BTW, I made a little research on source code and could not find anything related to my question. Perhaps, it's not implemented. Best regards, Grigoriy -- С уважением, Григорий Пузанкин -- _ -- 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 Hello Grigoriy, i think that you can access the information you need by using the dialplan function CALLERID(num-plan). It should contain the lower 7 bits of the Q.931 type-of-number/numbering-plan-identification octet. Best regards Gianluca -- _ -- 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] Call Pickup how to display CND of incoming number
On Tue, 2013-02-19 at 02:05 +, Klaverstyn, David C wrote: Is it possible to display the incoming calling number on a handset when trying to pick up a call from another handset? I currently have Call Pickup working using *8, I have also used the PickUp application successfully but I’m not sure how to use these features so the handsets show the incoming calling number and not the number that you have dialled to pick up the call. Regards David Klaverstyn Try setting sendrpid to pai in sip.conf -- Ishfaq Malik i...@pack-net.co.uk Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552 -- _ -- 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] PRI Called Party Number Info
I need to get type of called number (TON), which is displayed in pri debug messages: Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ] Does anyone know how to do it? According to documentation it is only possible for calling number. But I need to make decision in dialplan upon the value of type of called number. BTW, I made a little research on source code and could not find anything related to my question. Perhaps, it's not implemented. You did not specify for which version of Asterisk. I am assuming at least v1.8. I think you are wanting the CALLEDTON channel variable. It is set for incoming ISDN calls to the lower 7 bits of the Q.931 type-of-number/numbering-plan octet. The CALLERID(dnid-num-plan) should have the same value but I cannot find any code setting the value. (Looks like I missed setting that value. :)) I could only find the value being set to the CALLEDTON channel variable. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transcoding issues with siren14
On 28/02/2013, at 6:08 PM, Richard Kenner ken...@gnat.com wrote: Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and slin16 to slin32. So why is slin used as the intermediate instead of slin16? Do you have transcode_via_sln set in asterisk.conf? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- 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] blacklist caller ID
On 03/14/13 06:04, Geoff Lane wrote:
On Thursday, March 14, 2013, Joseph wrote:
Can someone refresh my memory how to backlist caller ID in
asterisk 1.8?
I had it working in ver. 1.4 but in 1.8 it changed.
I'm still using 1.4. In that I add a number to the blacklist with
CLI database put blacklist 0123456789 1
That is to add the number to the blacklist 'group' and give it a value
of 1.
Then is extensions.conf:
exten = main,n,GotoIf($[${BLACKLIST()}]?banned,1)
;(...)
; Blacklisted numbers
exten = banned,1,Playback(silence/2im-sorry)
exten = banned,n,Playback(cannot-complete-as-dialed)
exten = banned,n,Playback(privacy-you-are-blacklisted)
exten = banned,n,Playback(goodbye)
exten = banned,n,Hangup()
Hopefully, the same thing works in 1.8 and 10 (I'd be interested to
know because I intend to upgrade my hardware soon and will probably
take the opportunity to update * to a later version).
HTH,
--
Geoff
I got it. In asterisk 1.8 see:
http://www.voip-info.org/wiki/view/Asterisk+func+BLACKLIST
There is no need to give value of 1, it is just:
CLI database put blacklist 0123456789
--
Joseph
--
_
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[asterisk-users] AGI
Hi everybody, Does any one knows how to place a call from a shell agi? I guess is something like echo Exec Dial(DAHDI/g2/2010,,W). Algo how i get the dnid variable? 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] chan_mobile
El 12/03/13 19:11, Hans Witvliet escribió: Hi Emiliano, thanks for your reply, I think i might use it for a different project, I got an huawei-E1820 But at the moment i have to look at something else: The issue is contacting people not currently in the office. I've been trying to accomplish secure voice with a softphone through a vpn-tunnel, but the choosen softphone turns out less reliable then expected. While still working on that thread, other option is to equip each laptop with a_proper_ blue tooth dongle, and use their dumb/smart phone as an USB-audio device. If they are near their laptop, presence should allow me to use chan_mobile. (with an additional advantage not having to pay GSM-providers abroad) So, main issue is stability, reliability and usability for end users. Unless i can use a huawei as a single-channel BTS, i'll have to stick to use a BT-dongle. Hans Hans. I don't know if you use some time a bluetooth dongle. they sucks. If you can make your proyect will find anothers problems like disconections or poor sound or latency. Take a look at chipset of the BT dongles maybe you find something in your country. Best regards ! Emiliano. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 Interno 4 Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -- _ -- 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] Transcoding issues with siren14
Do you have transcode_via_sln set in asterisk.conf? No, but as I said in a later email, I found the problem: when computing the cost of a path, any downconvert has the same cost. So siren14 - slin - slin32 is the same cost as siren14 - slin16 - slin32 which is wrong. I fixed this by adding the magnitude of the difference in the sampling rate to the cost, but I'm not sure if that's the right solution. -- _ -- 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-Linux and DAHDI-Tools 2.6.3-rc1 Now Available
The Asterisk Development Team has announced the first release candidate of: DAHDI-Linux-2.6.3-rc1 DAHDI-Tools-2.6.3-rc1 dahdi-linux-complete-2.6.3-rc1+2.6.3-rc1 This beta release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-tools http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete - Fixes compile issue with dahdi-tools in Fedora 17 - Fixes compile issue with dahdi-linux when building OSLEC from the kernel source Issues closed in this release: DAHTOOL-60 DAHLIN-317 Shortlog of changes since v2.6.2: Russ Meyerriecks (1): Kbuild: Fix OSLEC build error Shaun Ruffell (1): build_tools/make_version: Only strip 'v' if followed by a digit. The diffstat from the v2.6.2 release: build_tools/make_version | 2 +- drivers/dahdi/Kbuild | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) For a full list of changes in these releases, please see the shortlog at: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/tags/v2.6.3-rc1 http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog;h=refs/tags/v2.6.3-rc1 Issues found in this release can be reported in the DAHDI-Linux [1] and DAHDI-Tools [2] projects at https://issues.asterisk.org/jira [1] https://issues.asterisk.org/jira/browse/DAHLIN [2] https://issues.asterisk.org/jira/browse/DAHTOOL Thank you for your continued support of 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
[asterisk-users] ERROR: Unknown signalling method ss7
Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. Now i`m unable to load chan_dahdi and libss7: myserver*CLI module load chan_dahdi.so ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 'ss7' at line 37. myserver*CLI module load libss7.so Unable to load module libss7.so Command 'module load libss7.so' failed. [Mar 14 22:30:05] WARNING[10124]: loader.c:423 load_dynamic_module: Module 'libss7.so' did not register itself during load [Mar 14 22:30:05] WARNING[10124]: loader.c:878 load_resource: Module 'libss7.so' could not be loaded. what is the problem? Can you please help me to solve this problem? Here is my config files: system.conf: = span=1,1,0,ccs,hdb3 bchan=1-15,17-31 mtp2=16 #dchan=16 loadzone = us defaultzone = us == chan_dahdi.conf: === [trunkgroups] [channels] callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes ;General options usecallerid = yes hidecallerid = no callwaiting = yes threewaycalling = yes transfer = yes echocancel = yes echocancelwhenbridged = yes rxgain = 0.0 txgain = 0.0 switchtype = national group = 1 signalling = ss7 ss7type = itu linkset = 1 ss7type = itu linkset = 1 pointcode = adjpointcode = defaultdpc = cicbeginswith = 1 channel = 1-15 cicbeginswith = 17 channel = 17-31 sigchan = 16 == Best 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
Re: [asterisk-users] ERROR: Unknown signalling method ss7
I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. Now i`m unable to load chan_dahdi and libss7: myserver*CLI module load chan_dahdi.so ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 'ss7' at line 37. myserver*CLI module load libss7.so Unable to load module libss7.so Command 'module load libss7.so' failed. [Mar 14 22:30:05] WARNING[10124]: loader.c:423 load_dynamic_module: Module 'libss7.so' did not register itself during load [Mar 14 22:30:05] WARNING[10124]: loader.c:878 load_resource: Module 'libss7.so' could not be loaded. what is the problem? Can you please help me to solve this problem? libss7-trunk cannot be used with any released version of Asterisk. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI
(A more specific subject may yield better answers -- better bait == better
fish.)
On Thu, 14 Mar 2013, Gustavo Salvador wrote:
Does any one knows how to place a call from a shell agi? I guess is
something like echo Exec Dial(DAHDI/g2/2010,,W).
While you can write an AGI in any language that has reasonable
STDIN/STDOUT facilities (fortran was a little bit frustrating), a shell
language like bash would not be my first choice.
Is there any reason you are not open to using a more 'mainstream'
scripting language like Perl or PHP? You'll find more people with that
experience willing to help.
My personal favorite is to use C because I know it best and because it is
a compiled language so your AGIs execute 'instantaneously.'
Whatever language you choose, use an established library for the AGI
protocol. Nobody gets it right the first time.
Algo how i get the dnid variable?
This implies that you need to read up a bit on the AGI protocol. The
second* thing an AGI should to is read the AGI variables -- one of which
is agi_dnid.
You can pass command line parameters to your AGI when you execute the agi()
function in your dialplan. For example:
exten = *,n,agi(my-first-agi,${AGENT-ID})
Your AGI can then access the command line arguments as you would
normally expect for a program executed from a shell -- $argv[1],
$argv[2], etc for PHP.
Personally, I dislike 'positional' parameters. I prefer to use the
getopt_long() function so my dialplan is 'self-documenting' and easier
to maintain so for me, the same command would look like:
exten = *,n,agi(my-first-agi,--agent-id=${AGENT-ID})
The difference is not all that apparent with a single parameter, but if
you have a half-dozen, it is much more obvious and once you've make the
change, you will always use it for consistency.
*) The first thing an AGI should do is set up a signal handler to catch
'SIGHUP' if the channel is hung up prematurely.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
--
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Re: [asterisk-users] ERROR: Unknown signalling method ss7
On 03/14/2013 11:04 PM, mohsen feyzzadeh wrote: Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. In case the 12 in Fedora 12 was not a typo, you do realize that Fedora 12 has been end-of-line for years and has more security holes than Swiss cheese? It makes sense to upgrade to the latest version of Fedora (which is 18) or switch to CentOS 6.4 which is more suited for server applications. You may also want to look at the latest versions of DAHDI (2.6.2/2.6.3rc) and Asterisk (11.2.1) assuming both work with an appropriate version of libss7. Regards, Patrick -- _ -- 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] Disagreements between codec_siren14 and Polycom sources
There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back correctly with the Digium's siren14 codec. I know the parameters are correct because the file is the same size as that made by the Digium codec. Both sets of decoders/encoders (Digium and Polycom/ITU) are symmetric and can decode what they encode, but neither can read the encoding of the other. Is there some subtle difference between G.722.1C and Siren14? -- _ -- 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] Call blocking issue
sorry if this is already asked? May get any help about call blocking issue Asterisk Version : Asterisk 1.6.2.7 FreePBX Version : 2.7.0.10 OS: CentOS My asterisk is running basic incomming and outgoin call is working but I can not configure call blocking and IVR. any one can give me some hitns? Thanks Jewel -- _ -- 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
