[asterisk-users] (no subject)
-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi, Im trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001 aors=2001 direct_media=no rtp_symmetric=yes force_rport=yes allow=alaw allow=speex allow=speex16 allow=speex32 allow=gsm [2001] type=aor qualify_frequency=5000 authenticate_qualify=yes max_contacts=1 remove_existing=yes [2001] type=auth auth_type=userpass password=test username=test Best Regards, Madushan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On 9 Feb 2015, at 15:32, Francisco Leonardo Mota francisco.m...@rnp.br wrote: Submission. Thanks, Uh, no problem?.. Steve -- _ -- 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] (no subject)
Submission. Thanks, Francisco Leonardo Mota Analista de Operações DAGSer - Diretoria Adjunta de Gestão de Serviços RNP – Rede Nacional de Ensino e Pesquisa Site:http://www.rnp.br Tel.:+55 61 3243-4384 Cel.:+55 61 9189-6660 -- _ -- 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] (no subject)
If you're using a redhat based distro, have you checked SELinux? Try disabling (will require a server reboot) Regards Ish On 3 September 2014 20:41, Steve Edwards asterisk@sedwards.com wrote: For future reference, a well chosen subject will yield more relevant replies. Better bait == better fish. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik 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, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW 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
[asterisk-users] (no subject)
Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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] (no subject)
Asterisk is not started. Start asterisk or look at the logs if there is any issues . Try asterisk -vvvgc and debug From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anthony Azzopardi Sent: Wednesday, September 03, 2014 11:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] (no subject) Hello asterisk-users, Just compiled and installed 11.12.0 however when I try to connect with rasterisk I get: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) It seems that asterisk.ctl is not created. -- _ -- 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] (no subject)
Did you start the Asterisk server? jg -- _ -- 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] (no subject)
For future reference, a well chosen subject will yield more relevant replies. Better bait == better fish. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Dahdi on Archlinux I was able to compile the latest 2.9 Dahdi in archlinux on the Beaglebone black without errors. I ran make install and make config. It installed the modules etc correctly but did not create an init script in systemd or anywhere else. Has anyone else been able to get dahdi to run in archlinux? How is the start script created? If I run dahdi_config it gives an error that /dev/dahdi.ctl does not exist. Doug -- _ -- 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] (no subject)
Hi, all I use Ubuntu 12.04.01 TLS and install asterisk 11.7.0 (tar.gz downloaded from asterisk.org). We named it Asterisk11. I want to generate a call file to /var/spool/asterisk/outgoing. This call will dial out to Local Channel and return to some Extens. Then Asterisk11 will generate a CDR records to MySQL's cdr table(in database mydatabase) via cdr_adaptive_odbc. The SIP/A221 is another asterisk machine named it Elastix24. I have two BIG QUESTIONs about cdr_adaptive_odbc. First, I have answered call from Elastix24 and I can listen the music file played from Asterisk11. In another word, this call should be answered and its billsec is greater than 0. Second, if I don't want to use forkcdr(), how to config it and I can get another cdr record that call from SIP/A221(Elastix24) to my Exten:77? I know that the outgoing file will make a call to Local Channel and try to Dial SIP/A221. If it answered, this old channel should be hangup and generate another new channel to connect to Extension:77(my callback exten). I can't find two cdr records in mycdr table. mysql select * from gvl_cdr; +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel | dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-+--+-+---+-+---+---+-+--+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:37:01 | | |77 | from-internal-out-7 | Local/77@from-internal-out-7-;2 | SIP/A221- | Dial| SIP/A221/77,30 | 17 | 0 | ANSWERED| 3 | | | 1389163021.1 | 1389163021.0 | 1| | 77 | | 7 | Even I try to add ForkCDR or ResetCDR. The billsec is 0 in other record(the 3th one). mysql select * from gvl_cdr; +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | calldate| clid | src | dst | dcontext| channel| dstchannel| lastapp | lastdata | duration | billsec | disposition | amaflags | accountcode | userfield | uniqueid | linkedid | sequence | peeraccount | phoneno | callerid | userid | +-++-+---+-++---+-+---+--+-+-+--+-+---+--+--+--+-+-+--++ | 2014-01-08 14:34:04 || | 77| from-internal-out-7 | Local/77@from-internal-out-7-;2| SIP/A221- | Dial| SIP/A221/77,30| 15 | 0 | ANSWERED|3 | | | 1389162844.1 | 1389162844.0 | 1| | 77 | | 7 | | 2014-01-08 14:34:04 | device 1000| 1000| 77| from-6 | Local/77@from-internal-out-7-;1| | ForkCDR | | 20 | 5 | ANSWERED|3 | | | 1389162844.0 | 1389162844.0 | 0| | 77 | | 7 | | 2014-01-08 14:34:24 | device 77 | 77 | 77| from-6 | Local/77@from-internal-out-7-;1| | Read| CALLBACK,custom-gvl/2,1,s,1,3 |0 | 0 | NO ANSWER |3 | | | 1389162844.0 | 1389162844.0 | 3| | | | 0 | - /var/spool/asterisk/outgoing/77.call Channel:Local/77@from-internal-out-7 WaitTime:30 Context:from-6 Extension:77 Priority:1 Set:CLID= Set:EXT=77 Set:USERID=7 -- /etc/asterisk/extensions.conf lists below:
[asterisk-users] (no subject)
To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=1 rtpend=10100 And My FW is setup to forward all incoming ports of range 1-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas Kellens jonas.kell...@telenet.be Subject: Re: [asterisk-users] RTP port ranges To: Andrew Colin and...@vsave.co.za Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5232dfc7.2030...@telenet.be Content-Type: text/plain; charset=iso-8859-1; Format=flowed Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! 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
[asterisk-users] (no subject)
Hi I am running following asterisk installed with apt on Debian 7.1. dpkg -l |grep asterisk ii asterisk 1:1.8.13.1~dfsg-3+deb7u1 amd64Open Source Private Branch Exchange (PBX) ii asterisk-config1:1.8.13.1~dfsg-3+deb7u1 all Configuration files for Asterisk ii asterisk-core-sounds-en-gsm1.4.22-1 all asterisk PBX sound files - en-us/gsm ii asterisk-modules 1:1.8.13.1~dfsg-3+deb7u1 amd64loadable modules for the Asterisk PBX If the incoming INVITE has the following two multiple bodies then it would not respond to that. It won't even send a Trying. We are using* TCP *only. Content-Type: application/sdp Content-Type: application/ISUP;base=itu-t92+;version=itu-t92+. Is this is a known issue? Are later version of asterisk able to deal with such multi-bodies INVITE? I got to play early media so it needs to make some sense out of first SDP. Best regards, Adnan -- _ -- 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] (no subject)
thanks for your response
with the code below i can't get the extenssions 223
exten = 529,1,Answer()
exten =
529,n,MixMonitor(test_num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
i can get my number only with uniqueid
test_num-0661xx_name-_529_UID-1376564701.1204.wav
any help please
thanks and regards
2013/8/13 Positively Optimistic positivelyoptimis...@gmail.com
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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[asterisk-users] (no subject)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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Re: [asterisk-users] (no subject)
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to record all the inboud
calls with the code below my question how can i do to save alse the sip
extenssion 223
exten = 529,1,Answer()
exten = 529,n,MixMonitor(test_${UNIQUEID}.wav|av(0)V(0))
exten = 529,n,Dial(SIP/223)
exten = 529,n,Hangup()
thanks and regards
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[asterisk-users] (no subject)
hello all, i want to have ooh323 connection between asterisk and cisco. in my scenario, asterisk is gateway and cisco is gatekeeper. this is my ooh323.conf file: [general] port=1720 bindaddr=192.168.0.227 gateway=yes faststart=yes h245tunneling=yes h323id=g...@test.com settracelevel=10 gatekeeper=192.168.0.212 context=from-trunk disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 with this config, gateway is registered in cisco gatekeeper correctly. but when i want to call from it, cisco reject my gateway and h225 asn1 messages say incomplete address. i searched a lot and understand that, if a cisco router acts as gateway, it sends h323-id as well as dialed number for gatekeeper but my gateway(which is asterisk), only send dialed number. therefore cisco gatekeeper doesn't know how route this call and reject it. if i define e164 number in ooh323.conf file, every thing is ok and call routed correctly. my question is: can asterisk work with cisco gatekeeper just by h323-id? if yes, how i can do this? in the other words, is it necessary to define e164 number in ooh323.conf file to have a correct connection or not? thanks in advance SAM -- _ -- 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] (no subject)
unsubscribe -- _ -- 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] (no subject)
Basic Dial Plan Why is this plan not engaging the line exten = 105,n,Dial(SIP/voipvoip.com/1703501) and dialing the 703 number? The logs and debug dont show any problems [incoming] exten = 44,1,Answer() exten = 44,n,Wait(1) exten = 44,n,Playback(beep) exten = 44,n,Goto(105,105,1) ; ; [105] exten = 105,1,Wait(2) exten = 105,n,Playback(hello-world) exten = 105,n,Dial(SIP/voipvoip.com/1703501) exten = 105,n,Hangup() -- _ -- 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] (no subject)
On Friday 12 April 2013, Thomas Perron wrote: Basic Dial Plan Why is this plan not engaging the line exten = 105,n,Dial(SIP/voipvoip.com/1703501) and dialing the 703 number? The logs and debug dont show any problems [incoming] exten = 44,1,Answer() exten = 44,n,Wait(1) exten = 44,n,Playback(beep) exten = 44,n,Goto(105,105,1) ; ; [105] exten = 105,1,Wait(2) exten = 105,n,Playback(hello-world) exten = 105,n,Dial(SIP/voipvoip.com/1703501) exten = 105,n,Hangup() Have you included the [105] context within the default context for the extension from which you are dialling 105? If 44 from the outside world is failing to trigger it, then it's possible that Asterisk is seeing the first 105 in Goto(105,105,1) as a priority rather than a context,extension,priority. Rename the [105] context to start with a letter. -- AJS Answers come *after* questions. -- _ -- 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] (no subject)
check this out http://msnbc.msn.com-report6.us/finance/-- _ -- 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] (no subject)
Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter Please help. Regards Akhilesh -- _ -- 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] (no subject)
On Monday 30 July 2012, akhilesh chand wrote: Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter It sounds as though you need to recompile DAHDI-Linux. (Did you compile it before you acquired this card?) Just download the latest DAHDI package Source Code, and compile and install it. If you didn't compile your own kernel from Source Code, then you will also need the package kernel-devel (on Fedora / CentOS) or linux-headers (on Ubuntu). -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- 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] (no subject)
Thanks ajs On Monday, July 30, 2012, A J Stiles wrote: On Monday 30 July 2012, akhilesh chand wrote: Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter It sounds as though you need to recompile DAHDI-Linux. (Did you compile it before you acquired this card?) Just download the latest DAHDI package Source Code, and compile and install it. If you didn't compile your own kernel from Source Code, then you will also need the package kernel-devel (on Fedora / CentOS) or linux-headers (on Ubuntu). -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- 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] (no subject)
http://goo.gl/XTjqx-- _ -- 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] (no subject)
http://adamdavidson-design.com/wp-content/themes/FastTrack/rogsfv.html?ncs=mmyq.jjsjss=sys.jyscjn=gyhp-- _ -- 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] (no subject)
Generate $500 $2500 a month - Own Your Own Business http://parkovani-u-letiste-praha.cz/httpwagregerw2.php?aforcamp=329 Well, this is it, Capet. kevon wingate Wed, 16 May 2012 18:07:05 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
http://aiscjmi.com/modules/mod_wdbanners/time.php?html143 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Karim Mardhani karim at vertexcommunication.ca
http://lists.digium.com/mailman/listinfo/asterisk-users wrote:
* Hi everyone,** ** I am trying to get Meetme to return back to the context
from where it** joined the meetme. For example a user uses the following
context to join a** conference, once user hangs up I would like to continue
executing the rest** of the dialplan. But when caller hangs up from the
conference I see on CLI** that meetme exited with non-zero status but none of
the rest of the** dialplan is executed. Please help. I am using asterisk
1.6.2.20** ** [default]** exten = _,1,MeetMe(1000,1pdMX)** exten =
_,n,noop(returned from meetme) ;After user hangs up should** come here**
exten = _,n,SoftHangup(${ORIG_CALLER})** exten =
_,n,SoftHangup(${CONF_CALLER})** exten = _,n,Hangup** exten =
h,1,noop(default-end)** exten = h,n,SoftHangup(${ORIG_CALLER})** exten =
h,n,SoftHangup(${CONF_CALLER})** exten = h,n,Hangup*
That's not how Asterisk works. When the caller hangs up, execution of
the current dialplan extension stops, and control passes to the 'h'
extension, if one exists in the current context.
Any processing you want to do when the caller hangs up must be done
in the 'h' extension. Cheers
Thanks Tony for the quick response. As you would see I have the h
extension defined but execution doesn't go to that either.
Tony
--
Tony Mountifield
Work: tony at softins.co.uk
http://lists.digium.com/mailman/listinfo/asterisk-users -
http://www.softins.co.uk
Play: tony at mountifield.org
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Karim Mardhani
Vertex Communication Ltd.
18667552554 ext. 103
www.vertexcommunication.ca
sip: ka...@sip2.vertexcommunication.ca
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Re: [asterisk-users] (no subject)
Hi sam,
Have solved the problem with your advice. Call drop in 10 seconds without
disconnecting a-party call. Thank you very much.
[TB]
exten =_X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten =_X.,4,Set(mainLoop=0)
;exten =_X.,5,Set(TIMEOUT(absolute)=5)
exten =_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,6,Dial(DAHDI/7/
09501032209,100,L(3[:1][:3000])g)
exten =_X.,7,Noop(${DIALEDTIME})
exten =_X.,8,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Cheers
Vinod Dharashive
Sent from BlackBerry® on Airtel
-Original Message-
From: Sam Govind govoi...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 7 Sep 2011 11:53:33
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] (no subject)
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Re: [asterisk-users] (no subject)
See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
; limit calls to ex-girlfriend to 300 seconds
exten = 123,1,Set(TIMEOUT(absolute)=300)
exten = 123,2,Dial(${EX-GIRLFRIEND})
exten = T,1,Playback(im-sorry)
exten = T,2,Playback(vm-goodbye)
exten = T,3,Hangup( )
Also see if Dial() command options L(x:y:z), or S(x) work out for you when
combined with option g.
On Wed, Sep 7, 2011 at 7:42 AM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is
disconnect both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten = _X.,4,Set(mainLoop=0)
exten = _X.,5,Set(TIMEOUT(absolute)=10); set time in milliseconds
exten = _X.,6,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,7,Dial(DAHDI/7/
09501032209,10,S(60))
exten = _X.,8,Noop(${DIALEDTIME})
exten =_X.,9,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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[asterisk-users] (no subject)
Hi team,
I am trying to find solution to hangup b-party call after 1 min with out
disconnecting the call of a-party but following dial plan which is disconnect
both the calls.
Please suggest me the solution.
[TB]
exten = _X.,1,Wait(${INCOMING_WAIT})
exten =_X.,2,Verbose(TB)
exten =_X.,3,Answer()
exten = _X.,4,Set(mainLoop=0)
exten = _X.,5,Set(TIMEOUT(absolute)=10) ; set time in milliseconds
exten = _X.,6,Playback(/var/callagent/prompts/monitor/thanks)
exten = _X.,7,Dial(DAHDI/7/
09501032209,10,S(60))
exten = _X.,8,Noop(${DIALEDTIME})
exten =_X.,9,Goto(TB,_X.,1)
exten =_X.,n,Hangup()
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
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[asterisk-users] (no subject)
We are having several issues with call parking in Asterisk 1.8.5. First, when a call is parked it is announcing the park location to the caller rather than the callee. We also are experiencing an issue whereby if you attempt to retrieve a parked call when a new call is incoming the new caller and the parked caller are connected together. -- _ -- 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] (no subject)
Miki Sent on my BlackBerry® from Vodafone -- _ -- 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] (no subject)
Sent on my BlackBerry® from Vodafone -- _ -- 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] (no subject)
Good morning gentlemen, is my first post in the list, now I'm starting asterisk
wanted to have your help for some questions.
Well the first function is as follow me. Here
I will demonstrate how this configuration follow me on my
extensions.conf but it is not working, and do not know why, but
something is missing?
You must set up followme.conf ?
What
I want is that the follow-me is enabled for any of the extensions
within the same context, like if I am absent from my table and go to
extension 2801 DataCenter where I need to spend all afternoon and I will
have the extension 2820 which enabled me to follow this extension and after
back to my desk withdraw follow me.
; Ativa Siga-me incondicional
[sigame-on]exten = _*71*.,1,NoCDR()
exten = _*71*.,2,Set(DB(CF/${CALLERID(num)})=${EXTEN:4})
exten = _*71*.,3,Playback(call-fwd-unconditionalforextension)
exten = _*71*.,4,SayDigits(${CALLERID(num)})
exten = _*71*.,5,Playback(is-set-to)
exten = _*71*.,6,SayDigits(${EXTEN:4})
exten = _*71*.,7,Playback(vm-saved)
exten = _*71*.,8,Playback(beep)
exten = _*71*.,9,Hangup
; Desativa o siga-me incondicional
[sigame-off]exten = _*72*,1,NoCDR()
exten = _*72*,2,DBdel(CF/${CALLERID(num)})
exten = _*72*,3,Playback(cancelled) exten = _*72*,4,Playback(beep)
exten = _*72*,5,Hangup
Bom, agora vamos ao pulo do gato, esse passo é muito importante pois é
ele quem verifica se existe ou não o siga-me para o ramal.
Vamos ao contexto:
[disca]
exten = _3XXX,1,Noop(CF/${EXTEN})
exten = _3XXX,2,Set(siga=${DB(CF/${EXTEN})})
exten = _3XXX,3,Dial(SIP/${siga},30,Ttw)
exten = _3XXX,4,Dial(SIP/${EXTEN}) ; Unconditional forward
exten = _3XXX,5,Hangup
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Re: [asterisk-users] (no subject)
you running GSM FWTs with asterisk ? On Mon, Apr 25, 2011 at 6:51 AM, Abid Saleem abid_aster...@hotmail.comwrote: HI, I am trying to setup a Class 4 termination setup using a kind of channel hunting scenerio. I have some SIP DID numbers assigned from the local telecom provider for termination. MY call comes from my wholesale client and lands on a switch, then it is routed to asterisk. I want asterisk to route this call to my local DID provider on the next available channel with DID number as the new Caller ID. This is just like GSM gateway that recieves the call and then re-originates the call using the next available SIM card number. Can someone help me how can I configure Asterisk to perform this? Thanks Abid. -- _ -- 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 -- Regards: (Muhammad υѕмαη ) -- _ -- 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] (no subject)
HI, I am trying to setup a Class 4 termination setup using a kind of channel hunting scenerio. I have some SIP DID numbers assigned from the local telecom provider for termination. MY call comes from my wholesale client and lands on a switch, then it is routed to asterisk. I want asterisk to route this call to my local DID provider on the next available channel with DID number as the new Caller ID. This is just like GSM gateway that recieves the call and then re-originates the call using the next available SIM card number. Can someone help me how can I configure Asterisk to perform this? Thanks Abid. -- _ -- 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] (no subject)
http://i-wikisport.com/product.php?page=32a -- _ -- 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] (no subject)
Anyone going to remove this spammer/scammer? 2010/12/19 Dmitry Kupchinetsky dkupchinet...@hotmail.com: http://www.barenakedbabies.com/shop/images/images.html -- _ -- 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] (no subject)
http://www.barenakedbabies.com/shop/images/images.html -- _ -- 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] (no subject)
Hi, I want to know that I have created a IAX2 trunk between two trunk I am observing a packet rate of 50packet/sec mean packetization time=20ms but I want to know that how to change the packetization time I have placed trunk freq=50 in general section of IAX but can not see any difference and its still working on 20ms thanks in advance for help Regards Ali Raza Anjum-- _ -- 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] (no subject)
Hi, Does anyone know where this is suddenly coming from? -- _ -- 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] (no subject)
On Sat, Oct 16, 2010 at 4:35 PM, Dan Journo d...@keshercommunications.comwrote: Hi, Does anyone know where this is suddenly coming from? -- _ -- 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 Where what is suddenly coming from? Cheers - The Mick -- _ -- 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] (no subject)
Jjo Thanks, Jeff Jones mailto:jeff.jjo...@gmail.com tel:12489068232 mobile:12486323130 -- _ -- 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] (no subject)
Hi, I have a problem with a SIP trunk between Asterisk and central OXE Alcatel, especially sometimes are not received inbound calls with following messages: -- Executing [...@test:1] AGI(SIP/800-084250f8, agi://127.0.0.1/test.agi) in new stack -- AGI Script agi://127.0.0.1/test.agi completed, returning 0 == Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN' I configured the sip.conf file: [800] type=peer host=172.XX.XX.XX username=test secret=XXX insecure=very context=test disallow=all allow=alaw allow=ulaw and the extensions.conf file: exten = 375,1,AGI(agi://127.0.0.1/test.agi) I attach to this email the sip messages receveid by Asterisk when the problem occurs. Thanks for your help. Best regards, GP --- SIP read from 172.25.51.1:10011 --- INVITE sip:3...@172.24.10.188;user=phone SIP/2.0 Supported: replaces,100rel User-Agent: ABS GW v5.1.0 P-Asserted-Identity: ISDN_T2 sip:+521776...@mercury Content-Type: application/sdp To: sip:3...@172.24.10.188;user=phone From: ISDN_T2 sip:+521776...@mercury;tag=cc01ff37a60521d35da001f98edda0ac Contact: sip:172.25.51.1 Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 CSeq: 684819861 INVITE Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c Max-Forwards: 70 Content-Length: 314 v=0 o=OXE 1279704517 1279704517 IN IP4 172.25.51.1 s=abs c=IN IP4 172.25.51.4 t=0 0 m=audio 32712 RTP/AVP 8 0 4 97 a=sendrecv a=rtpmap:8 PCMA/8000 a=ptime:20 a=maxptime:30 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:30 a=rtpmap:4 G723/8000 a=ptime:30 a=maxptime:30 a=rtpmap:97 telephone-event/8000 - --- (13 headers 17 lines) --- Sending to 172.25.51.1 : 5060 (no NAT) Using INVITE request as basis request - 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 Found peer '800' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 4 Found RTP audio format 97 Peer audio RTP is at port 172.25.51.4:32712 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G723 for ID 4 Found audio description format telephone-event for ID 97 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0xd (g723|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 172.25.51.4:32712 Looking for 375 in sedoc (domain 172.24.10.188) list_route: hop: sip:172.25.51.1 --- Transmitting (no NAT) to 172.25.51.1:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1 From: ISDN_T2 sip:+521776...@mercury;tag=cc01ff37a60521d35da001f98edda0ac To: sip:3...@172.24.10.188;user=phone Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 CSeq: 684819861 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:3...@172.24.10.188 Content-Length: 0 -- Executing [...@sedoc:1] AGI(SIP/800-084250f8, agi://127.0.0.1/mercury.agi) in new stack -- AGI Script agi://127.0.0.1/mercury.agi completed, returning 0 == Auto fallthrough, channel 'SIP/800-084250f8' status is 'UNKNOWN' Scheduling destruction of SIP dialog '82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1' in 32000 ms (Method: INVITE) --- Reliably Transmitting (no NAT) to 172.25.51.1:5060 --- SIP/2.0 603 Declined Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c;received=172.25.51.1 From: ISDN_T2 sip:+521776...@mercury;tag=cc01ff37a60521d35da001f98edda0ac To: sip:3...@172.24.10.188;user=phone;tag=as3455cb36 Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 CSeq: 684819861 INVITE ser-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ccsedoc*CLI --- SIP read from 172.25.51.1:10011 --- ACK sip:3...@172.24.10.188;user=phone SIP/2.0 Call-ID: 82f9cd2d2e6c712d7528ef88802ba...@172.25.51.1 From: ISDN_T2 sip:+521776...@mercury;tag=cc01ff37a60521d35da001f98edda0ac To: sip:3...@172.24.10.188;user=phone;tag=as3455cb36 Via: SIP/2.0/UDP 172.25.51.1;branch=z9hG4bK3e9645a733e59941286088d7a4945d8c CSeq: 684819861 ACK Content-Length: 0 - -- _ -- 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] (no subject)
Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption is right. How can I check if those calls are being sent with caller ID. Because all I see on console output for the phone call is this -- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27 instead) -- Nobody picked up in 1000 ms -- Hungup 'DAHDI/56-1' It doesn't show where it actually tried to dial or not. I know it works because if I sent it to the in house number it calls that number and if someone answers it they get the person who is on hold in the queue. It only fails on outside the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -- _ -- 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] (no subject)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James A. Shigley Sent: Friday, July 16, 2010 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] (no subject) Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption is right. How can I check if those calls are being sent with caller ID. Because all I see on console output for the phone call is this -- Channel 0/8, span 3 got hangup, cause 50 (sometimes cause 1 or 27 instead) -- Nobody picked up in 1000 ms -- Hungup 'DAHDI/56-1' It doesn't show where it actually tried to dial or not. I know it works because if I sent it to the in house number it calls that number and if someone answers it they get the person who is on hold in the queue. It only fails on outside the building calls. So where do I check to see if it is or isn't attaching caller ID. Let's assume I'm right and the CID is the issue; What config and/or context do I need to change so that the when a queue tries to place a call to an agent there is caller ID? James Shigley -- 1. obviously it did dial, otherwise you wouldn't get nobody picked up 2. in your dialplan, put this line before queue Exten = 1,1,Set(CALLERID(num)=201212) - change 1,1 to context appropriate values and 201212 to a proper DID for your location. Do these this a post a CLI output with verbose set to 5 or higher. -- _ -- 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] no subject
Hello, list. I've set up an outbound alerting system to play a recording when systems go down, etc. and I'm noticing that cellphones tend to answer() and then start ringing the actual handset. So far, I've verified this behavior with Verizon, T-Mobile, and Google Voice (the last produces a SERIOUS delta between bogus answer and actual answer). Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? In the short term, I just have the call play MOH for ten seconds before announcing that all hell has broken loose in the server room, but it¹d be nice to have something a bit more accurate and reliable. Cheers, 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] no subject
On Fri, Jul 9, 2010 at 12:57 PM, Mike Ely mike...@amyskitchen.net wrote: Has anyone figured out how to detect the actual cellphone answer rather than the bogus one sent by the cell carrier? *CLI core show application AMD -- 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
[asterisk-users] (no subject)
http://leyvacrystaljd.blog23.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] (no subject)
-- 祝您愉快！！ Aaron Chen 陈江涛 -- _ -- 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] (no subject)
On Fri, Mar 19, 2010 at 3:13 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Fail2ban is a must. I was a victim of such attacks, and have implemented some other measures too, but fail2ban is a must have with the link posted by Matt which describes how to set it up for asterisk. Make sure you put your own ip address in ignore list otherwise it can block you too. You may also consider to use BFD (Brute Force Detection) [1] as your tool for log analysis. We have a detailed tutorial [2] on how to install and configure BFD, using Asterisk rules [3] for SIP and IAX protocols. Our approach is not to use iptables but to block the communication with the attacker using route del -host $ATTACK_HOST reject. To unban a specific IP we will use a manual command like route del -host $ATTACK_HOST reject. This is not probably not the best method but it works for us till now. Best regards, Ioan. [1] - http://www.rfxn.com/projects/brute-force-detection/ [2] - http://www.modulo.ro/Modulo/ro/Articole/Securitate_pentru_servere_Asterisk.html [3] - http://www.modulo.ro/Modulo/downloads/tools/tenora.bfd.tar.gz -- _ -- 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] (no subject)
Hello, I'm looking for some advice on securing Asterisk. Recently my servers been under several brute-force SIP attacks. I have several remote sites, as well as many roaming users, who may have PC softclients and/or SIP based hardphones. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? I haven't looked at Secure SIP in quite a while, is that now integrated into 1.6 ? One thing that's confusing me in my config, is that I thought that if I set NAT=no in sip.conf, then I wouldn't be able to connect to that SIP account unless I was on the local LAN, specified by locallan= However in some testing, I'm finding that I can still connect from an external SIP client. Also, I tried setting one SIP account from host=dynamic to host=ipaddr, and when that client tried to register, then Asterisk complained that the account wasn't supposed to be trying to register. My next step is also to upgrade my Asterisk itself up to the latest stable 1.6 Any other suggestions? Thanks, Adrian -- _ -- 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] (no subject)
On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asterisk. Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] (no subject)
On Fri, 19 Mar 2010, Adrian Marsh wrote: I’m looking for some advice on securing Asterisk. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Asterisk doesn't differentiate between a hard phone and a soft phone. You can restrict by IP address for soft phones as well. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? I'm a 1.2 Luddite, so I can't speak for 1.6. I think any brute force or DOS security policy needs to be implemented external to Asterisk. I don't think there are any AMI events you could listen to. I think you are limited to what you can scrounge out of a log file. How about setting up a couple of honey-pot SIP accounts with obvious passwords and in the context fire off a user event? Then you could listen for the event via AMI. Any other suggestions? Repost with a meaningful subject -- a blank subject labels you as a newbie who is probably not worth the time of members with relevant experience. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Fail2ban is a must. I was a victim of such attacks, and have implemented some other measures too, but fail2ban is a must have with the link posted by Matt which describes how to set it up for asterisk. Make sure you put your own ip address in ignore list otherwise it can block you too. On 2010-03-18 8:45 PM, Matt Riddell li...@venturevoip.com wrote: On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asteri... Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] (no subject)
Please descard me from the asterisk users list...thanks (Abu Nasar Mahmud) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
If you read your message all the way to the end, and every posting, you will discover exactly how to do that on your own. asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users nasar mahmud wrote: Please descard me from the asterisk users list...thanks (Abu Nasar Mahmud) Checked by AVG - www.avg.com Version: 9.0.733 / Virus Database: 271.1.1/2660 - Release Date: 01/31/10 14:35:00 -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi, That model HP or Dell server that I recommend for a TE412P card for about 200 users? Thank you very much. _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
All, I want to know if it's possible to create a log file per context? and each time a context is restarted a ne x log file is created. regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
After doing a little research on this, the answer is a limited yes. Asterisk has 6 logging files to be used. If you aren't using all 6, you could designate any unused files to a context and use the log application to feed that specific log file. Since you would be doing this in a custom fashion, you could manually roll that log with a system command at the top of the context. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mickael ropars Sent: Tuesday, October 20, 2009 3:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] (no subject) All, I want to know if it's possible to create a log file per context? and each time a context is restarted a ne x log file is created. regards Mickael ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Tue, 20 Oct 2009, mickael ropars wrote:
I want to know if it's possible to create a log file per context? and
each time a context is restarted a ne x log file is created.
This is not clear to me. Contexts are not restarted. What are you trying
to log?
Asterisk has the system() application which will execute any arbitrary
Linux command line so you can do pretty much anything.
Asterisk doesn't have the native ability to create log files as I think
you described. How would you handle 2 calls entering the same context at
effectively the same time? There are race conditions to consider both
for file creation and writing.
Maybe this will give you some ideas:
[wildcard-test]
exten = _!,1, verbose(1,[${CONTEXT}:${EXTEN}])
exten = _!,n, system(logger -i -p local0.info -t
${CONTEXT} ${CALLERID(num)} entered context)
exten = _!,n, answer()
exten = _!,n, hangup()
exten = _x,4, playback(demo-congrats)
exten = _x,n, system(logger -i -p local0.info -t
${CONTEXT} ${CALLERID(num)} finished)
exten = _x,n, hangup()
exten = h,2,system(logger -i -p local0.info -t
${CONTEXT} ${CALLERID(num)} hung up)
exten = h,n,hangup()
This will log every entry to the context to syslogd. You can configure
syslogd (/etc/syslog.conf) to separate the log entries as desired.
This is pretty inefficient -- it creates at least 4 processes (2 on entry,
2 on hangup) for every call.
I had an application several years ago that required logging how long each
caller was in each context. I used resetcdr(w) and enhanced
cdr_addon_mysql.c. When the call finished, I executed an AGI that added up
the cdrs and rated the call.
If you post questions with meaningful subject lines, you may attract the
interest of someone who has solved your exact problem and you make it
easier for the next guy to research.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] (no subject)
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[asterisk-users] (no subject)
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service , temporary unavailable .,
what to do to solve this problem
In other words how to stop MOH since asterisk detect 183 and even if i can
do that when the 183 comes from my soft switch which will allow user to hear
the Ring Back Tone
i found in the app_dial.c
case AST_CONTROL_RINGING:
Thanks in advance
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[asterisk-users] (no subject)
I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
- ameu...@yahoo.fr wrote: I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. I tried looking for some answers based upon your subject but nothing came up. This may be what you're looking for: http://lmgtfy.com/?q=asterisk+java+api --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On 19 Mar 2009, at 15:08, ameu...@yahoo.fr wrote: I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. Ok. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
use ami http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+Java or Ajam http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM) 2009/3/19 ameu...@yahoo.fr I have to develop a VoIP application. I need to know how to use Java APIs to communicate to my client application with asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Asterisk user number: 1099 Linux user: #443184 shazaum.googlepages.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
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Re: [asterisk-users] (no subject)
Right On Mon, Feb 23, 2009 at 9:07 PM, Lê Văn Hòa ho...@inet.vn wrote: ko gui nua -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
ko gui nua -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hello, I have problem after killall -9 asterisk and asterisk -f Asterisk stops to send to DNS resolving of trunks Regards -- We never did too much talking anyway So don't think twice, it's all right ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
V 1.4 When I do a show channels I get the following. CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 2 active channels 2 active calls I need to kill these SIP channels, but the only thing I have found when searching is the soft hangup solution - which simply doesn't do anything to these channels. CLI soft hangup SIP/7110-b495d3b0 CLI soft hangup SIP/7110-afd286e0 CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 2 active channels 2 active calls Can someone suggest a better way of getting rid of these channels? My solution so far has been to restart Asterisk... not a good solution. Thanks Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
What asterisk cli shows when you soft hangup these channels Shariq On Fri, Sep 5, 2008 at 11:55 PM, Bill Andersen [EMAIL PROTECTED]wrote: V 1.4 When I do a show channels I get the following. CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 2 active channels 2 active calls I need to kill these SIP channels, but the only thing I have found when searching is the soft hangup solution - which simply doesn't do anything to these channels. CLI soft hangup SIP/7110-b495d3b0 CLI soft hangup SIP/7110-afd286e0 CLI show channels Channel Location State Application(Data) SIP/7110-b495d3b0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 SIP/7110-afd286e0[EMAIL PROTECTED]:2Up Page(Local/[EMAIL PROTECTED]Local/71 2 active channels 2 active calls Can someone suggest a better way of getting rid of these channels? My solution so far has been to restart Asterisk... not a good solution. Thanks Bill ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi All, I have one doubt, suppose we have conference between 3 users (PCM companded voice channels) then we add the streams together with scaling but data which a user can receive will include his own voice information also or i think we should substract his info. from the combined data, also as the total sum of scaling factors should be 1 how we decide these scaling factors becoz these factors decides audio gain of each channel? Can you plz suggest me steps to follow to implenent voice conference using DSP(I am using Fixed point DSP TMS320c55x) and Components to use from DSP and level of buffering for incoming data. Thanks in advance. Rahul jadhav. Rahul Jadhav Junior Design Engineer Spectross Digital System (P) Ltd. No. 4, Siri Fort Road | New Delhi - 110049 Phone: +91-9990865914 | 011-26264077 Email : [EMAIL PROTECTED] Web : www.spectross.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I try to build Asterisk this is the error I'm getting. src/add.c:1: error: CPU you selected does not support x86-64 instruction set I just can't seem to find what i need to set to get this to build. Thanks _ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_WL_Refresh_messenger_video_072008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Hi - I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I try to build Asterisk this is the error I'm getting. src/add.c:1: error: CPU you selected does not support x86-64 instruction set You may not have the right sources for your kernel. You may have the 32-bit sources instead of the 64-bit ones. What kind of CPU is it? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi I m making a call from one asterisk server to an asterisk client The call gets completed but I want it to send dtmf signals The dialplan I have made for this is like exten = 205,1,Answer exten = 205,n,Wait(15) exten = 205,n,Playback(dtmf-1) exten = 205,n,Wait(20) but it does not send any dtmf signal where is the problem?? CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS*** ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Use SendDTMF. --- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote: From: Neha Punia [EMAIL PROTECTED] Subject: [asterisk-users] (no subject) To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Date: Thursday, July 3, 2008, 10:29 AM Hi I m making a call from one asterisk server to an asterisk client The call gets completed but I want it to send dtmf signals The dialplan I have made for this is like exten = 205,1,Answer exten = 205,n,Wait(15) exten = 205,n,Playback(dtmf-1) exten = 205,n,Wait(20) but it does not send any dtmf signal where is the problem?? CAUTION - Disclaimer * This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS End of Disclaimer INFOSYS***___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
But if I m using this SendDTMF it does not send anything
I m using it like this in extension.conf
exten = 205,1,Answer
exten = 205,n,Wait(20)
exten = 205,n,Playback(dtmf-1)
exten = 205,n,Wait(20)
exten = 205,n,SendDTMF(9)
exten = 205,n,Wait(5)
exten = 205,n,Read(digito)
exten = 205,n,SayDigits(${digito})
exten = 205,n,Hangup
on the console it only shows tht the call completed and no message about the
DTMF and in the log files it shows like :
Jul 3 17:21:01 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Setting NAT on RTP to 0
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Outgoing Call for 205
Jul 3 17:21:27 DEBUG[896] chan_sip.c: (Provisional) Stopping retransmission
(but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Acked pending invite 102
Jul 3 17:21:27 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:21:27 DEBUG[896] chan_sip.c: build_route: Contact hop: sip:[EMAIL
PROTECTED]
Jul 3 17:21:47 DEBUG[896] chan_sip.c: * Detected inband DTMF '1'
Jul 3 17:22:18 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '205'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'default'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'SIP/3001-008d8ce0'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'Hangup'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:21:27'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '2008-07-03 17:22:23'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '56'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '56'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'ANSWERED'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is 'DOCUMENTATION'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '1215085887.0'
Jul 3 17:22:23 DEBUG[896] pbx.c: Function result is '(null)'
Jul 3 17:22:23 DEBUG[896] chan_sip.c: update_call_counter(205) - decrement
call limit counter
Jul 3 17:22:23 NOTICE[896] pbx_spool.c: Call completed to SIP/3001
Jul 3 17:22:23 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 103: Match Found
Jul 3 17:22:24 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'
Jul 3 17:23:57 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:24:09 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'
Jul 3 17:25:47 DEBUG[896] chan_sip.c: Stopping retransmission on '[EMAIL
PROTECTED]' of Request 102: Match Found
Jul 3 17:25:54 DEBUG[896] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]'
It says detected inband dtmf 1 but says nothing about 9.
Am I doing anything wrong in the extension.conf.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob
Sent: Thursday, July 03, 2008 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] (no subject)
Use SendDTMF.
--- On Thu, 7/3/08, Neha Punia [EMAIL PROTECTED] wrote:
From: Neha Punia [EMAIL PROTECTED]
Subject: [asterisk-users] (no subject)
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com
Date: Thursday, July 3, 2008, 10:29 AM
Hi
I m making a call from one asterisk server to an asterisk
client
The call gets completed but I want it to send dtmf signals
The dialplan I have made for this is like
exten = 205,1,Answer
exten = 205,n,Wait(15)
exten = 205,n,Playback(dtmf-1)
exten = 205,n,Wait(20)
but it does not send any dtmf signal
where is the problem??
CAUTION - Disclaimer *
This e-mail contains PRIVILEGED AND CONFIDENTIAL
INFORMATION intended solely
for the use of the addressee(s). If you are not the
intended recipient, please
notify the sender by e-mail and delete the original
message. Further, you are not
to copy, disclose, or distribute this e-mail or its
contents to any other person and
any such actions are unlawful. This e-mail may contain
viruses. Infosys has taken
every reasonable precaution to minimize this risk, but is
not liable for any damage
you may sustain as a result of any virus in this e-mail.
You should carry out your
own virus checks before opening the e-mail or attachment.
Infosys reserves
[asterisk-users] (no subject)
Hello everybody I have configures asterisk server and i am using TE220P digium card. Here is the content of the /etc/zaptel.conf file ### span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone = in defaultzone = in the content of /etc/asterisk/zapata.conf is as follow [channels] context=incoming switchtype=national ;pridialplan=national usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callprogress=no callerid=asreceived group=1 channel=1-15,17-31 # output of zttool is as follow #9474; Alarms Span #9474; #9474; RED T2XXP (PCI) Card 0 Span 1 #9474; OK T2XXP (PCI) Card 0 Span 2 #9474; Output of cat /prox/zaptel/1 is as follow Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS RED 1 TE2/0/1/1 Clear (In use) RED 2 TE2/0/1/2 Clear (In use) RED 3 TE2/0/1/3 Clear (In use) RED 4 TE2/0/1/4 Clear (In use) RED 5 TE2/0/1/5 Clear (In use) RED 6 TE2/0/1/6 Clear (In use) RED 7 TE2/0/1/7 Clear (In use) RED 8 TE2/0/1/8 Clear (In use) RED 9 TE2/0/1/9 Clear (In use) RED 10 TE2/0/1/10 Clear (In use) RED 11 TE2/0/1/11 Clear (In use) RED 12 TE2/0/1/12 Clear (In use) RED 13 TE2/0/1/13 Clear (In use) RED 14 TE2/0/1/14 Clear (In use) RED 15 TE2/0/1/15 Clear (In use) RED 16 TE2/0/1/16 HDLCFCS (In use) RED 17 TE2/0/1/17 Clear (In use) RED 18 TE2/0/1/18 Clear (In use) RED 19 TE2/0/1/19 Clear (In use) RED 20 TE2/0/1/20 Clear (In use) RED 21 TE2/0/1/21 Clear (In use) RED 22 TE2/0/1/22 Clear (In use) RED 23 TE2/0/1/23 Clear (In use) RED 24 TE2/0/1/24 Clear (In use) RED 25 TE2/0/1/25 Clear (In use) RED 26 TE2/0/1/26 Clear (In use) RED 27 TE2/0/1/27 Clear (In use) RED 28 TE2/0/1/28 Clear (In use) RED 29 TE2/0/1/29 Clear (In use) RED 30 TE2/0/1/30 Clear (In use) RED 31 TE2/0/1/31 Clear (In use) RED I am new to asterisk and googled around , configured the asterisk server. Now when i make a call from outside , it give me busy tone.. and when i call from softphone .. it shows me as show below -- Executing [EMAIL PROTECTED]:1] Dial(SIP/bikrish-09b21980, Zap/g1/600833) in new stack [Jul 3 19:14:34] WARNING[6018]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/bikrish-09b21980' status is 'CONGESTION' I am not able to figure out the problem. Any kind of help would be appericiated. Thanking you bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P As for your problem looks like you are trying to use the wrong span for dial out. On Thu, Jul 3, 2008 at 8:50 AM, Bikrish Amatya [EMAIL PROTECTED] wrote: Hello everybody I have configures asterisk server and i am using TE220P digium card. Here is the content of the /etc/zaptel.conf file ### span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,2,0,ccs,hdb3 bchan=32-46,48-62 dchan=47 loadzone= in defaultzone = in the content of /etc/asterisk/zapata.conf is as follow [channels] context=incoming switchtype=national ;pridialplan=national usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no callprogress=no callerid=asreceived group=1 channel=1-15,17-31 # output of zttool is as follow #9474; Alarms Span #9474; #9474; RED T2XXP (PCI) Card 0 Span 1 #9474; OK T2XXP (PCI) Card 0 Span 2 #9474; Output of cat /prox/zaptel/1 is as follow Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 HDB3/CCS RED 1 TE2/0/1/1 Clear (In use) RED 2 TE2/0/1/2 Clear (In use) RED 3 TE2/0/1/3 Clear (In use) RED 4 TE2/0/1/4 Clear (In use) RED 5 TE2/0/1/5 Clear (In use) RED 6 TE2/0/1/6 Clear (In use) RED 7 TE2/0/1/7 Clear (In use) RED 8 TE2/0/1/8 Clear (In use) RED 9 TE2/0/1/9 Clear (In use) RED 10 TE2/0/1/10 Clear (In use) RED 11 TE2/0/1/11 Clear (In use) RED 12 TE2/0/1/12 Clear (In use) RED 13 TE2/0/1/13 Clear (In use) RED 14 TE2/0/1/14 Clear (In use) RED 15 TE2/0/1/15 Clear (In use) RED 16 TE2/0/1/16 HDLCFCS (In use) RED 17 TE2/0/1/17 Clear (In use) RED 18 TE2/0/1/18 Clear (In use) RED 19 TE2/0/1/19 Clear (In use) RED 20 TE2/0/1/20 Clear (In use) RED 21 TE2/0/1/21 Clear (In use) RED 22 TE2/0/1/22 Clear (In use) RED 23 TE2/0/1/23 Clear (In use) RED 24 TE2/0/1/24 Clear (In use) RED 25 TE2/0/1/25 Clear (In use) RED 26 TE2/0/1/26 Clear (In use) RED 27 TE2/0/1/27 Clear (In use) RED 28 TE2/0/1/28 Clear (In use) RED 29 TE2/0/1/29 Clear (In use) RED 30 TE2/0/1/30 Clear (In use) RED 31 TE2/0/1/31 Clear (In use) RED I am new to asterisk and googled around , configured the asterisk server. Now when i make a call from outside , it give me busy tone.. and when i call from softphone .. it shows me as show below -- Executing [EMAIL PROTECTED]:1] Dial(SIP/bikrish-09b21980, Zap/g1/600833) in new stack [Jul 3 19:14:34] WARNING[6018]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) == Auto fallthrough, channel 'SIP/bikrish-09b21980' status is 'CONGESTION' I am not able to figure out the problem. Any kind of help would be appericiated. Thanking you bikrish ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Thu, 3 Jul 2008, Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? I would say the main thing that is needed is a grammar and spelling checker, followed by some degree of nominal assessment of conceptual integrity and coherence. The latter may be impossible to implement, but the former would be beneficial. But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? I would say the main thing that is needed is a grammar and spelling checker, followed by some degree of nominal assessment of conceptual integrity and coherence. The latter may be impossible to implement, but the former would be beneficial. But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) I'm trilingual, but English is by far my best language. If I had to write a post on a technical mailing list in one of the other languages, I would certainly take the time to ensure that it sounds reasonably coherent. I cannot fault people for poor/limited English. But there is a difference between someone who tried and someone who didn't, and it is reflected in the overall level of culture that comes across in the substance of their post, the formulation of their thoughts, and so on. Somebody that *both* speaks/writes English poorly -- *and* uses incomprehensible, Philistine gibberish (excuse me, AOLer short-hand) -- deserves what they earn. There seems to be a remarkable coincidence of these two proclivities as often as not. -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: Steve Edwards wrote: On Thu, 3 Jul 2008, Alex Balashov wrote: C F wrote: The number one skill for setting up asterisk is learn how to communicate since it's a communication application :P Oh, if only more newbie posters on this list would heed that advice. ) How about rejecting emails that don't have a subject? ) How about rejecting top posted replies? ) How about rejecting posts to -dev until the poster's account is more than a couple of days old? Or until they've earned a couple of karma points? Or a challenge/response confirming this post is about changing the C source code? I would say the main thing that is needed is a grammar and spelling checker, followed by some degree of nominal assessment of conceptual integrity and coherence. The latter may be impossible to implement, but the former would be beneficial. But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) I'm trilingual, but English is by far my best language. If I had to write a post on a technical mailing list in one of the other languages, I would certainly take the time to ensure that it sounds reasonably coherent. I cannot fault people for poor/limited English. But there is a difference between someone who tried and someone who didn't, and it is reflected in the overall level of culture that comes across in the substance of their post, the formulation of their thoughts, and so on. Somebody that *both* speaks/writes English poorly -- *and* uses incomprehensible, Philistine gibberish (excuse me, AOLer short-hand) -- deserves what they earn. There seems to be a remarkable coincidence of these two proclivities as often as not. -- Alex Bilingual, Trilingual, -lingual does not necessarily include English as one of the languages. It is for some a great effort just trying to write in English, never mind the effort of knowing colloquialism, etc. So not being fluent, not being able to be as coherent as a native English speaker would, does not make me or someone else eligible for an answer. No wonder so many think that monolingual people with English as their only language are arrogant Yes, diatribes and flames are accepted //Peter ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Fri, 4 Jul 2008, Peter Lindquist wrote: Steve Edwards wrote: But deciphering posts from our non-English-speaking members is half the challenge/fun :) Seriously though, good for them for trying. I wouldn't. What are you if you speak 3 languages? Trilingual. What are you if you speak 2 languages? Bilingual. What are you if you only speak 1 language? American :) Bilingual, Trilingual, -lingual does not necessarily include English as one of the languages. It is for some a great effort just trying to write in English, never mind the effort of knowing colloquialism, etc. So not being fluent, not being able to be as coherent as a native English speaker would, does not make me or someone else eligible for an answer. No wonder so many think that monolingual people with English as their only language are arrogant Yes, diatribes and flames are accepted Boy, did you miss the mark. I am a monolingual American. I was giving non-English-speakers props for trying and poking fun at myself and my countrymen. Lighten up. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Alex Balashov wrote: ) How about rejecting emails that don't have a subject? That is an excellent idea. If a person doesn't have enough clue to use a subject, then we're really just feeding the beast when we indulge the question with an answer. And the archived version of that question/answer are pretty useless, too. Thx. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi : asterisk didn't send voice message to my mail([EMAIL PROTECTED]).My main configured files are: extensions.conf: [from-pstn] exten = 9711315,1,Dial(SIP/3000,30) exten = 9711315,2,VoiceMail([EMAIL PROTECTED]) exten = 9711315,3,PlayBack(vm-goodbye) exten = 9711315,4,HangUp() sip.conf: [3000] type=friend username=3000 secret=1234567 host=dynamic context=from-pstn [EMAIL PROTECTED] voicemail.conf: [ff_tutorial] 3000 = 1234567,3000,[EMAIL PROTECTED] And these are in console: Jun 29 12:08:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Accepting call from '3322000' to '9711315' on channel 0/1, span 1 Jun 29 12:08:15 DEBUG[24207]: chan_zap.c:1554 zt_enable_ec: Enabled echo cancellation on channel 1 -- Executing Dial(Zap/1-1, SIP/3000|30) in new stack Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:1889 create_addr_from_peer: Setting NAT on RTP to 0 Jun 29 12:08:15 DEBUG[24257]: chan_sip.c:2085 sip_call: Outgoing Call for 3000 -- Called 3000 Jun 29 12:08:16 DEBUG[24203]: chan_sip.c:1468 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/3000-08941d28 is ringing Jun 29 12:08:16 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 3 on channel Zap/1-1 Jun 29 12:08:21 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:31 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:41 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 -- Nobody picked up in 3 ms Jun 29 12:08:47 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication -1 on channel Zap/1-1 Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:2450 sip_hangup: update_call_counter(3000) - decrement call limit counter Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1392 __sip_ack: Acked pending invite 102 Jun 29 12:08:47 DEBUG[24257]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 29 12:08:47 DEBUG[24257]: app_dial.c:1661 dial_exec_full: Exiting with DIALSTATUS=NOANSWER. -- Executing VoiceMail(Zap/1-1, [EMAIL PROTECTED]) in new stack Jun 29 12:08:47 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'vm-intro' (language 'en') Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 29 12:08:47 DEBUG[24203]: chan_sip.c:1415 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Not Found Jun 29 12:08:51 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:52 DEBUG[24257]: app.c:1235 ast_lock_path: Locked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX' Jun 29 12:08:52 DEBUG[24257]: app.c:1256 ast_unlock_path: Unlocked path '/var/spool/asterisk/voicemail/ff_tutorial/3000/INBOX' Jun 29 12:08:52 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'beep' (language 'en') Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:08:53 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals -- Recording the message Jun 29 12:08:53 DEBUG[24257]: app.c:568 ast_play_and_record_full: play_and_record: None, /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9, 'wav49|gsm|wav' Jun 29 12:08:53 DEBUG[24257]: app.c:585 ast_play_and_record_full: Recording Formats: sfmts=wav49 -- x=0, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav49, 0x88b0f48 -- x=1, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: gsm, 0x88b12a0 -- x=2, open writing: /var/spool/asterisk/voicemail/ff_tutorial/3000/tmp/GE15v9 format: wav, 0x88b15e0 Jun 29 12:08:53 DEBUG[24257]: chan_zap.c:4981 zt_indicate: Requested indication 18 on channel Zap/1-1 Jun 29 12:09:01 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:09:11 DEBUG[24257]: chan_zap.c:3669 zt_handle_dtmfup: DTMF digit: # on Zap/1-1 -- User ended message by pressing # Jun 29 12:09:11 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 160 sample intervals -- Playing 'auth-thankyou' (language 'en') Jun 29 12:09:11 WARNING[24207]: chan_zap.c:8093 zt_pri_error: Call Reference Length not supported: 0 Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:09:12 DEBUG[24257]: channel.c:1777 ast_settimeout: Scheduling timer at 0 sample intervals Jun 29 12:09:12
[asterisk-users] (no subject)
In the setup tutorial @ http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation it states the potential issue regarding setting up UniqueID as the primary key, but doesn't state how to rectify this? What is the proper way to make sure this is done right? Also, has anyone built a simple front end for non technical folk to utilize for accessing the data simply for overview when billing etc is not important (small company)? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
the subject of this thread has been on this list way too many times just search the archives. On 5/23/08, Joseph L. Casale [EMAIL PROTECTED] wrote: In the setup tutorial @ http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation it states the potential issue regarding setting up UniqueID as the primary key, but doesn't state how to rectify this? What is the proper way to make sure this is done right? Also, has anyone built a simple front end for non technical folk to utilize for accessing the data simply for overview when billing etc is not important (small company)? Thanks! jlc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Dear Steve, We have installed Asterisk with Digium card TE110P , install MFC R2 connect to PSTN (indonesia) using DIG13 MFCR2 siemens EWSD, Germany. asterisk working normaly, outgoing call ok, incoming call ok. but in central office /PSTN having SLA(service level alarm). If It happend, all channel blocked immedeately.Our Question : 1. What are the problem ,steve? 2. how we adjust configuration of mfcr2 to matched with mfcr2 pstn/telcom? 3.how to configure MFCR2 DID (incoming only) mode? 4.how to konfigure MFCR2 DOID(incomong and outgoing)mode ? thanks for your helps before bestregards, dini handayani Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
This may be more helpful as far as Asterisk implementation. Sorry I cannot be of more help, I have never dealt with this tech. http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:06 AM, Arthur [EMAIL PROTECTED] wrote: http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. Maybe if you ask the telco to turn off the SLA blocking. It may not solve the underlying issue but it may allow you to continue inbound and outbound without service interruption providing it does not drop any active calls as well. Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:12 AM, Steve Totaro [EMAIL PROTECTED] wrote: This may be more helpful as far as Asterisk implementation. Sorry I cannot be of more help, I have never dealt with this tech. http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 Thanks, Steve Totaro On Mon, Apr 28, 2008 at 9:06 AM, Arthur [EMAIL PROTECTED] wrote: http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. i always say if you pay your bills you should get the support you diserve. every provider is almost always willing to help out his clients if they express their needs with precision. one more thing : nothing compares to having a friend working at the providers company so get yourself one. Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. you shoudl add something like this to your base code .. if finish-email == 'yes': keyboard.hit(enter) else: keyboard.write(text) :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
On Mon, Apr 28, 2008 at 9:32 AM, Arthur [EMAIL PROTECTED] wrote: Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. i always say if you pay your bills you should get the support you diserve. every provider is almost always willing to help out his clients if they express their needs with precision. one more thing : nothing compares to having a friend working at the providers company so get yourself one. You are preaching to the choir. I have dealt with all the big and many of the small players here in the US. I always say people that do the right thing and work hard will be rewarded but more often than not, they are taken advantage of. This is not Utopia, these guys at the telcos are overworked, work in a monolithic bureaucracy, and many probably hate their jobs. They love to close tickets ASAP since that is how they are evaluated. As soon as I get a good helpful tech, I get their DID and praise the heck out of them (almost to the point of brown nosing) and CC their supervisor (with their permission of course). Normal support channels get me answers like we cannot do that, or we can but it will take about two weeks. Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. you shoudl add something like this to your base code .. if finish-email == 'yes': keyboard.hit(enter) else: keyboard.write(text) :) True, true, but coffee tends to stave off incomplete or incoherent postings. Sometimes I look at posting made at the end of the day or before the caffeine kicks in and they make no sense whatsoever :) Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Apparently, there is a SIP(diversionheader) field that fixes the problem below, but I cannot find any docs or examples of how to use it in my dialplan. Any help would be appreciated. We have a Cisco CallManager where users forward their numbers, so PSTN-PSTN calls get this error... -Greg --- SIP read from 209.253.136.204:5060 --- INVITE sip:[EMAIL PROTECTED];transport=UDP SIP/2.0 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY Supported: timer Accept: multipart/mixed,application/media_control+xml,application/sdp Max-Forwards: 9 Min-SE: 60 Contact: sip:[EMAIL PROTECTED]:5060;transport=UDP Content-Type: application/sdp Content-Length: 500 v=0 o=BroadWorks 31324769 1 IN IP4 209.253.136.204 s=- c=IN IP4 209.253.136.204 t=0 0 m=audio 24418 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=ptime:20 a=fmtp:101 0-15 a=fmtp:18 annexb=no a=x-cxc-sess:04c2e65cf9a2aa97-1 a=x-cxc-info:cGVlci1wdWI9NjQuMTk5LjUxLjIxMDtwZWVyLXNkcD0yMDkuMjUzLjEyOS4xNTc6MjYzMDY7 a=x-cxc-info:cGVlci1yb3V0ZS10YWc9aW50ZXJuYWw7YW5jaG9yLWRzdD0yMDkuMjUzLjEzNi4yMDQ6MjQ0MTg7 a=sendrecv - --- (14 headers 17 lines) --- Sending to 209.253.136.204 : 5060 (no NAT) Using INVITE request as basis request - [EMAIL PROTECTED] Found peer 'McLeodUSA' Found RTP audio format 18 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 209.253.136.204:24418 Found audio description format G729 for ID 18 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw| g729)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 209.253.136.204:24418 Looking for 9723814678 in default (domain 209.33.163.37) list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=UDP ns2*CLI --- Transmitting (no NAT) to 209.253.136.204:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.253.136.204:5060;branch=z9hG4bK-301-4807c9fe-a5bac2-5babdc2b;received=209.253.136.204 From: Cell Phone TXsip:[EMAIL PROTECTED];tag=501ff0a-13c4-4807c9fe-a5bac2-7f73fc7a To: CISTERA 9723814678sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/4693412073-08fdbf78, SIP/[EMAIL PROTECTED]) in new stack Audio is at 192.168.5.14 port 13374 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.5.10:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 17 Apr 2008 22:08:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 28662 28662 IN IP4 192.168.5.14 s=session c=IN IP4 192.168.5.14 t=0 0 m=audio 13374 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called [EMAIL PROTECTED] ns2*CLI --- SIP read from 192.168.5.10:49365 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.5.14:5060;branch=z9hG4bK4a44607b;rport From: Cell Phone TX sip:[EMAIL PROTECTED];tag=as178544f0 To: sip:[EMAIL PROTECTED];tag=16863906 Date: Thu, 17 Apr 2008 22:06:54 GMT Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Allow-Events: telephone-event Content-Length: 0 - --- (9 headers 0 lines) --- ns2*CLI --- SIP read from 192.168.5.10:6060 --- INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.5.10:6060;branch=z9hG4bK32426484 From: Cell Phone TX sip:[EMAIL PROTECTED];tag=16863908 To: sip:[EMAIL PROTECTED] Date: Thu, 17 Apr 2008 22:06:55 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 User-Agent: Cisco-CCM4.1 Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK CSeq: 101 INVITE Max-Forwards: 70 Remote-Party-ID: Cell Phone TX sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off Contact: sip:[EMAIL PROTECTED]:6060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 227 v=0 o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 192.168.5.10 s=SIP Call c=IN IP4
Re: [asterisk-users] (no subject)
On Fri, 2008-02-22 at 10:38 +0530, sandeep wrote:
for example:
dial to a extension(123).if the user didnot pick the call, caller
should get a ivr script(Enter 1 to to dial operator and 2 to go to
voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it
should go to the voicemail of calle's extension.
It's really pretty easy.
; Call the SIP peer, let the phone ring for 20 seconds
exten = 123,1,Dial(SIP/some_sip_peer,20)
; Play the press-1-or-press-2 prompt, get one digit
; from the caller, and save it to a variable called
; ${option}
exten = 123,n,Read(option,press-1-or-press-2,1)
; If the caller enters 1, send the call to the [some_context] context,
; to the operator extension, priority 1
exten = 123,n,GotoIf($[${option} = 1]?some_context,operator,1)
; Otherwise, send the call to voicemail
exten = 123,n,VoiceMail([EMAIL PROTECTED])
I haven't actually taken the time to test this in my own dialplan, but
it should work. Obviously you'll want to change the name of the SIP
peer you're dialing, as well as the location of the operator extension.
--
Jared Smith
Community Relations Manager
Digium, Inc.
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