Re: [asterisk-users] Dialing a call back out on same SIP trunk as it came in

I had in a same situation and solved by Background 1 sec. silence. On Wed, Nov 25, 2015 at 5:45 PM, Brian :: wrote: > add a pause in the dialplan for a second then proceed.. > > > > On Wed, Nov 25, 2015 at 2:27 PM, Tony Mountifield > wrote: > >> In article

Re: [asterisk-users] Calls not hanging up

Your call is up on VoiceMail you should check dialstatus before sending user to VoiceMail. On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain da...@vex.net wrote: This just started after upgrading to 11.11.0. After a call is completed (both ends hang up) the call still shows as active. #

Re: [asterisk-users] dialplan =how many concurrent calls

you can use GROUP and GROUP_COUNT n,Set(GROUP()=aname) n,GotoIf($[${GROUP_COUNT(aname)} 8]?${EXTEN},200) 200,Hangup On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com wrote: Hi guys. Does somebody knows how to get the concurrent calls from the dial plan? Or. How can

Re: [asterisk-users] Trunk issue

file is executable? can you show ls -l /var/lib/asterisk/agi-bin On Mon, Apr 28, 2014 at 7:12 PM, Haley,Scott A scott.ha...@edwardjones.comwrote: It runs but hangs with the output of: perl tbsdial.agi 81101 GET VARIABLE astexten Right now, it is a simple perl script. Here is the entire

Re: [asterisk-users] Trunk issue

if that is the case then check again Perl Asterisk AGI. On Mon, Apr 28, 2014 at 7:33 PM, Haley,Scott A scott.ha...@edwardjones.comwrote: One more thing. I have this exact same script working on an Asterisk 1.8 box. This is a new Asterisk 11.7 box. Thanks, Scott Haley 5-2244

Re: [asterisk-users] FW: clients unable to auth

Hello, Try this [6004] type=friend host=dynamic disallow=all allow=ulaw allow=alaw callerid=6004 Peter secret=XXX context=default port=9060 nat=force_rport,comedia deny=0.0.0.0 permit=IP/255.255.0.0,IP/255.255.0.0,PrvIP/255.255.0.0, PrvIP/255.255.0.0 On Wed, Apr 16, 2014 at

Re: [asterisk-users] Asterisk 1.8.24 : illegal instruction

Hello, you can check the asterisk binary with. file /usr/sbin/asterisk and linked library ldd /usr/sbin/asterisk On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens jonas.kell...@telenet.bewrote: On 20-11-13 14:43, A J Stiles wrote: On Wednesday 20 November 2013, Jonas Kellens wrote: Hello,

Re: [asterisk-users] outbound call issue

some more information's will help sort out the issue. On Fri, Oct 18, 2013 at 2:30 PM, shiva kumar sivakumar.kara...@gmail.comwrote: Dear All, i had an issue when we are going to call back the number from asterisk its ringing as the customer mobile is switched off. And also it

Re: [asterisk-users] MusicOnHold starts magically for no reason

if you don't use MOH just don't load module res_musiconhold.so On Fri, Oct 18, 2013 at 6:24 PM, Alban Elziere alban.elzi...@nevox.frwrote: Thank you for pointing this thread. So, looks like no solution exists to correct this (as I understand)... as it is part of the standard. Have you found

Re: [asterisk-users] What linux distro most popular for Asterisk

We are using Debian 32bit and 64bit on standalone and on VMs without any issue. On Thu, Oct 17, 2013 at 10:15 AM, Frederic Van Espen frederic...@gmail.comwrote: On 10/17/2013 09:47 AM, Alban Elziere wrote: I'm using Ubuntu server (32bit mainly), standalone or VM (esxi) with good stability.

Re: [asterisk-users] is g729 codec free? or under license???

it true? On Sat, Oct 12, 2013 at 1:08 PM, Asghar Mohammad asghar...@gmail.comwrote: HI, You don't need a g729 installed in pass throw mode. if both ends have codec g729 you can just enable on both peers. and asterisk should pass the codec from 1 end to other. but make sure you

Re: [asterisk-users] is g729 codec free? or under license???

HI, You don't need a g729 installed in pass throw mode. if both ends have codec g729 you can just enable on both peers. and asterisk should pass the codec from 1 end to other. but make sure you are not doing transcoding of any type answering the call playing voice prompts etc. On Sat, Oct 12,

Re: [asterisk-users] Capture Media IP in CDR

hi, you have not mentioned which cdr backend you are using. peer ip is saved in variable CHANNEL(peerip). if you are using mysql for cdr backend you can create a field in cdr table (field name can b any of your choice) in dialplan assign the value of CHANNEL(peerip) to you ip field and asterisk

Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

Hi, Bad boys trying to guess a valid username. in sip.conf uncomment alwaysauthreject=yes and Asterisk always reject 1st invite. On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 01/10/13 15:44, gincantalupo wrote: On Tue, Oct 1, 2013 at 5:07 AM,

Re: [asterisk-users] problem to get MWI working

HI Asmaa, I don't know how MWI works in Voicemail but as i understand it just create a .call file and put in /var/spool/asterisk/outgoing and asterisk execute that file. i am using similar method for sending fax to from email. i show you some examples from my php scripts. 1. in voicemail

Re: [asterisk-users] problem to get MWI working

Hi Asmaa, Have you enabled debug to console in logger.conf? enable debug in logger.conf console = notice,warning,error,debug and reload Asterisk. On Sun, Sep 29, 2013 at 4:48 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hi Asghar, Thanks a lot for your proposed solution! MWI is turned on

Re: [asterisk-users] iax: unable to transfer - one way audio

Hi, If you post your configuration someone may help you. On Sat, Sep 28, 2013 at 5:03 PM, Sean Darcy seandar...@gmail.com wrote: On 09/27/2013 09:08 PM, Sean Darcy wrote: We have zoiper connected over iax to asterisk in Sydney. The call is to asterisk in New York. The caller in NZ can hear

Re: [asterisk-users] How to bind to ipv4 ipv6

Hi, Please Search the List there is already a post and solution. On Fri, Sep 27, 2013 at 3:58 PM, Daniel van den Berg aster...@suretel.co.za wrote: Hi All, This is my 1st post so lets go. What I need to achieve is the following. I have server with both IPv4 addresses and IPv6 addresses.

Re: [asterisk-users] The call is established but without exchanged voice packets

Hello, If Asterisk version is 1.6 use nat=force_rport,comedia On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have set the direct media to be off, but still doesn't work. I am not sure about NAT configuration! SIP.conf, [general] section

Re: [asterisk-users] The call is established but without exchanged voice packets

Hello, i think your logic is wrong please explain me what are you trying to do? [internal] exten = 7002,1,Answer() exten = 7002,n,Playback(vm-nobodyavail) exten = 7002,n,Hangup() exten = 7001,1,Dial(SIP/7001,60) exten = 7001,n,Hangup() try this dial 7002 and you should listen vm-nobodyavail or

Re: [asterisk-users] The call is established but without exchanged voice packets

Hello, paste you extension context. On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have Asterisk 1.8.10.1 Moving to nat=force_rport,comedia hasn't solved the problem. Still having the same error! I am not sure if this is related to the problem here,

Re: [asterisk-users] Can't connect to Asterisk cli

remove content of /var/log/asterisk/messages /var/log/asterisk/messages run asterisk and post content of /var/log/asterisk/messages to pastebin. On Thu, Sep 19, 2013 at 9:39 AM, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, No, another installation haven't solved the problem! It looks

Re: [asterisk-users] sipgate outgoing calls

you have insecure=port,invite in sipgate peer? On Thu, Sep 19, 2013 at 12:26 PM, gpxctawjc...@irational.org wrote: On Thu, 19 Sep 2013, Miguel Oyarzo wrote: Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format

Re: [asterisk-users] Can't connect to Asterisk cli

become root sudo su - or su -l give your password. if asterisk is already running connect to asterisk -rvvvc otherwisw asterisk -c. if you want asterisk run as daemon asterisk and then connect to asterisk asterisk -rvvvc On Wed, Sep 18, 2013 at 2:13 PM, Asmaa Ahmed asabatg...@hotmail.com

Re: [asterisk-users] Can't connect to Asterisk cli

SELinux exists in Ubuntu? On Wed, Sep 18, 2013 at 2:45 PM, Ishfaq Malik i...@pack-net.co.uk wrote: Have you checked your SELinux settings? On 18 September 2013 13:13, Asmaa Ahmed asabatg...@hotmail.com wrote: Hello, I have started using Asterisk recently on my Ubuntu server. I

Re: [asterisk-users] Can't connect to Asterisk cli

i think you messed 2 installs of asterisk. if you compile asterisk from sources it not insert init script. you can test installing to /opt. 1. cd to asterisk sources folder 2. make distclean 3. ./configure --prefix=/opt/asterisk 4. make 5. sudo make install 6. /opt/asterisk/sbin/asterisk -c

[asterisk-users] Asterisk-1.8.23.1 mysql cdr

) disable all existing options usegmtime etc. added new cli option cdr mysql cdrzone. it will show you selected timezone. patch can be download from http://www.world-call-trade.com/asterisk/cdr_mysql_cdrzone.patch please report back here. BEST REGARDS Asghar Mohammad

[asterisk-users] Asterisk-1.8.23.1 mysql cdr

Hi list, I am using Asterisk1.6.2 form a long time and upgarding to Asterisk-1.8.23.1. I am using mysql backend for cdr. in asterisk-1.6.2 i have usegmtime=yes and it works as expected insert cdr date in GMT0. now i tested Asterisk-1.8.23.1 and asterisk-11.5 with same results no matter what i

Re: [asterisk-users] Asterisk 1.8 drop calls after 15 minutes

hi, it seems your vpn connection drop. is you vpn on WiFi of any other high latency network? On Tue, Sep 10, 2013 at 1:05 PM, Administrator TOOTAI ad...@tootai.netwrote: Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones

Re: [asterisk-users] No remote address on RTP instance - On Ringing

i have used a2billing some time ago maybe there is somthing new . you can try shoot up loglevel to 4 and see the verbose of agi that may give you some hint. On Tue, Sep 10, 2013 at 7:34 PM, jg webaccou...@jgoettgens.de wrote: Maybe the ringtone from downstream is not reaching asterisk,

Re: [asterisk-users] Cisco SPA303 won't ring for more than 60 seconds

sip set debug on and see trace of upload on pastebin. On Wed, Aug 21, 2013 at 8:25 PM, jg webaccou...@jgoettgens.de wrote: At first I also thought this might be a phone setting. But then I found the same 60s to be true for a variety of SIP phones (Snom, Cisco, ...), despite the 300s timeout

Re: [asterisk-users] Am I being hacked?

he, some bad boys trying to guess configured extensions. in sip config in general set alwaysauthreject = yes . in cli sip set debug on and watch ip and block in firewall, iptables. On Mon, Aug 19, 2013 at 7:50 PM, Ira i...@extrasensory.com wrote: Hello Steve, Sunday, August 18, 2013,

Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

just remove username. type peer authenticate by ip On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin and...@vsave.co.za wrote: change server two to host = dynamic then add register = on server 1 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote: Even I tried the type as friend.. but no

Re: [asterisk-users] G729 Passthrough How To

As my understanding Asterisk always pass-thu g729 if both ends have this codec. But if you answer the call or play some audio before dialing to end point then asterisk stay between both legs. In case of VM. you should install g729 if your prompts are in g729 format. As a2billing play voice prompts

Re: [asterisk-users] Asterisk trunking between two location

hi, paste server a trunk also, if you want register why you are not using host=dynamic? both servers are on 10.10.10.0 ? if no then check your deny permit seting. On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Also tried one more scenario, particularly

Re: [asterisk-users] Asterisk trunking between two location

. On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.comwrote: hi, paste server a trunk also, if you want register why you are not using host=dynamic? both servers are on 10.10.10.0 ? if no then check your deny permit seting. On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N

Re: [asterisk-users] Asterisk trunking between two location

, 2013 at 10:25 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: can't we use without register command both way as peer to peer? On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad asghar...@gmail.comwrote: 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b

Re: [asterisk-users] Asterisk trunking between two location

-27d2' Actually the trunk which i mentioned in my first email, it was working... and from today it is not Still breaking... what could be the reason... ! On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad asghar...@gmail.comwrote: yes you can. just create trunks on both side with static ip

Re: [asterisk-users] Asterisk trunking between two location

make a call and post cli log On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: still the peer shows unreachable let me restart and give a try... On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad asghar...@gmail.comwrote: *1st Location* [manila] type

Re: [asterisk-users] how to send dtmf after pause ?

hi, you can add more w (ww1234#) for more delay. On Fri, Jun 7, 2013 at 7:17 PM, Yves A. yves...@gmx.de wrote: This would be possible with an agi... the agi can wait for silence or 10 seconds, as u like and then play the dtmf tones and bridge the call to your extension afterwards.

Re: [asterisk-users] how to send dtmf after pause ?

hi. check here for agi http://forum.voxilla.com/threads/introducing-waits-w-in-dial-destination-number-variable.14628/ On Fri, Jun 7, 2013 at 7:50 PM, Sean Darcy seandar...@gmail.com wrote: On 06/07/2013 01:17 PM, Yves A. wrote: This would be possible with an agi... the agi can wait for

Re: [asterisk-users] Installing Asterisk 11 on VirtualBox: Illegal Instruction

what is host architecture ? try to install ubuntu x86 not x86_64. On Thu, Jun 6, 2013 at 5:12 PM, jorgeart...@protoboardmx.com wrote: I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two guides. The

Re: [asterisk-users] Difference MySQL between 1.6.x and 11.4.x

asterisk trying connect to mysql via socket remove that line from config files. 1 check if port 3306 is open in iptables on both servers. 2 check permissions on db for user Asterisk. On Mon, Jun 3, 2013 at 9:18 PM, Olivier CALVANO o.calv...@gmail.com wrote: on this server we don't have

Re: [asterisk-users] Not able to build the chan_sip.c module

please provide more information. how you are try to build asterisk, what is output of configure. witch headers configure script not found etc. On Tue, May 28, 2013 at 9:29 AM, upendra uppi...@gmail.com wrote: hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at

Re: [asterisk-users] DTMF recognized after call establishment

i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something

Re: [asterisk-users] DTMF recognized after call establishment

work around was block dtmf. set wrong type of dtmf in incoming trunk. On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar

Re: [asterisk-users] Registration timed out - for created users

you don't need register = string here, it only need you want asterisk register to another sip proxy as client. just remove that line and you should fine. for X-lite or any other sip phone the user AlphaUser is sufficient. On Fri, May 24, 2013 at 12:32 PM, luke devon luke_de...@yahoo.com wrote:

Re: [asterisk-users] Cut offs on outgoing SIP calls

asterisk is behind nat? On Wed, May 15, 2013 at 8:18 PM, Daniel - Asterisk earohua...@gmail.comwrote: Hello everyone, I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones show the gollowing messages when are being dropped:

Re: [asterisk-users] Cut offs on outgoing SIP calls

please show us peer configuration. On Wed, May 15, 2013 at 9:46 PM, Daniel - Asterisk earohua...@gmail.comwrote: Users (softphones) are behind a NAT, Asterisk has its own public ip address On Wed, May 15, 2013 at 1:30 PM, Asghar Mohammad asghar...@gmail.comwrote: asterisk is behind nat

Re: [asterisk-users] Cut offs on outgoing SIP calls

sip set debug peer 90102 and check in log why call drop or upload log somewhere. configuration seems ok. On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk earohua...@gmail.comwrote: Current configuration follows: [general] context=default allowguest=no alwaysauthreject=yes

Re: [asterisk-users] Monitoring SIP trunk status on call by call basis

i think DIALSTATUS is not suitable for failover if trunk is down you get dialstatus after time out in dial string. it is too late for failover, you can use some script to check if destination host is up. if you want to do failover when destination host is up then dialstatus are good. On Tue, May

Re: [asterisk-users] Using PHPMyAdmin to remotely access Asterisk MySQL Database

what problem you encounter? On Tue, May 14, 2013 at 9:42 PM, Lobna Hegazy lobna.heg...@gmail.comwrote: Dear All, I'm trying to connect to Asterisk CDR database using PHPMyAdmin but unfortunately all my trials and searches failed. So I'd be more than grateful if someone helped me

Re: [asterisk-users] Sangoma Wanpipe Driver

Dial(DAHDI/i0/number, is it not Dial(DAHDI/r0/number or Dial(DAHDI/R0/number or Dial(DAHDI/g0/number or Dial(DAHDI/G0/number? On Mon, May 13, 2013 at 12:53 PM, Yves A. yves...@gmx.de wrote: mmh... actually supportline is closed... why proceeds the call to dahdi/pseudo-?? i have never

Re: [asterisk-users] Integrate Astreisk with SIP interface

what you mean by interface? if you want connect sip phone with asterisk there are 2 file to modify 1. sip.conf 2. extensions.conf. for creating sip user add following in sip.conf [ivr_user] defaultuser=ivruser ;username for sip phone secret=ivruser ;password for sip phone

Re: [asterisk-users] time zone setting in asterisk

you can try to set usegmtime=no in cdr.conf On Sun, May 12, 2013 at 3:40 AM, Joseph syscon...@gmail.com wrote: Which file in Asterisk have a setting for time zone? When asterisk record incoming call in Master.csv the time is 6hr. ahead. I'm on: Canada/Mountain zone -- Joseph --

Re: [asterisk-users] time zone setting in asterisk

solved? On Sun, May 12, 2013 at 5:39 PM, Joseph syscon...@gmail.com wrote: On 05/12/13 12:18, Asghar Mohammad wrote: you can try to set usegmtime=no in cdr.conf I commented it out, as no is the default setting; but for some reason it was enabled on Gentoo installation. -- Joseph

Re: [asterisk-users] ISP trunk session ID?

exactly please? Thanks in Advance, Nick. On 5/10/13, Asghar Mohammad asghar...@gmail.com wrote: hi, you can try to change sip user agent and sdp session s , owner in sip config same as your phone,s (modem). asterisk by default send user agent = asterisk version , s= asterisk , o

Re: [asterisk-users] dahdi driver not getting install

install kernel source. On Sat, May 11, 2013 at 10:15 AM, Harish Mandowara hari...@cdac.in wrote: Dear, I have redhat enterprise linux 6.3. after uname -a i am getting Linux genesys-dell 2.6.32-279.el6.x86_64 #1 SMP Wed Jun 13 18:24:36 EDT 2012 x86_64 x86_64 x86_64 GNU/Linux now when i

Re: [asterisk-users] dahdi driver not getting install

installing kernel source on debian use *apt*-*get insatll* linux-headers-$( *uname* -r) On Sat, May 11, 2013 at 12:20 PM, Alec Davis siva...@paradise.net.nzwrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On

Re: [asterisk-users] dahdi driver not getting install

he is using debian. debian have yum? On Sat, May 11, 2013 at 2:44 PM, Andrew Colin and...@vsave.co.za wrote: Do a yum install kernel-devel kernel-headers Reboot and it will work Sent from my iPhone On 11 May 2013, at 12:20 PM, Alec Davis siva...@paradise.net.nz wrote:

Re: [asterisk-users] ISP trunk session ID?

hi, you can try to change sip user agent and sdp session s , owner in sip config same as your phone,s (modem). asterisk by default send user agent = asterisk version , s= asterisk , o= asterisk. some providers are not happy if they see asterisk word :) On Sat, May 11, 2013 at 12:27 AM, Sergej

Re: [asterisk-users] question about CDR

hi, asterisk insert cdr when call is hangup and last dial statment, i dont understatnd why you are using 2 dial statment on same extenstion? if you you want dial to both extensions you can use 506,1,Dial(SIP/223SIP/276) if you want dial both same time or if you want to do failover the check Dial

Re: [asterisk-users] Gateway?

are you talking about sip to pstn? thats called fxo ATA. On Tue, Apr 30, 2013 at 8:59 PM, Don Kelly d...@donkelly.biz wrote: Guys and gals - these are all excellent answers - I am not being clear, I think. ** ** Let me see if I can illustrate it. ** ** If you cannot see my

Re: [asterisk-users] h323-sip: one way connection

try UserByAlias=yes in general and type=user in user context. On Fri, Apr 26, 2013 at 9:48 AM, s m sam.gh1...@gmail.com wrote: oh yes, i'm using h323 not openh323 On Fri, Apr 26, 2013 at 1:16 AM, Asghar Mohammad asghar...@gmail.comwrote: nuFone h323 or openh323? On Thu, Apr 25, 2013

Re: [asterisk-users] h323-sip: one way connection

nuFone h323 or openh323? On Thu, Apr 25, 2013 at 9:33 PM, s m sam.gh1...@gmail.com wrote: flavor? i do not understand what you mean. please explain more. thanks On Wed, Apr 24, 2013 at 8:16 PM, Asghar Mohammad asghar...@gmail.comwrote: what flavor of h323 you are using? On Wed, Apr 24

Re: [asterisk-users] h323-sip: one way connection

what flavor of h323 you are using? On Wed, Apr 24, 2013 at 8:50 AM, s m sam.gh1...@gmail.com wrote: thanks Asghar, i do it, but no thing happened:( asterisk do not identify host line as ip address of the other end On Tue, Apr 23, 2013 at 9:15 PM, Asghar Mohammad asghar

Re: [asterisk-users] h323-sip: one way connection

the address by host=192.168.0.146 but asterisk can not find it? why? On Mon, Apr 22, 2013 at 8:23 PM, Asghar Mohammad asghar...@gmail.comwrote: please post cli output for both calls. On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote: hello everybody i want to have sip

Re: [asterisk-users] h323-sip: one way connection

please post cli output for both calls. On Mon, Apr 22, 2013 at 11:32 AM, s m sam.gh1...@gmail.com wrote: hello everybody i want to have sip connection between two asterisk systems (145 and 146). connection from 145 to 146 is ok but i can not call from 146 to 145. this is h323.conf file in

Re: [asterisk-users] External call control for Asterisk

AGI is your friend. check A2billing. On Fri, Apr 19, 2013 at 10:43 AM, Lenz Emilitri lenz.lo...@gmail.comwrote: Not sure if that's what you are looking for, but I would think about having the dialplan call a web service (maybe using CURL) and passing account and current number. The system

Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists

to fix it. please let me know if you have any other suggestion. thanks sam On 4/11/13, Asghar Mohammad asghar...@gmail.com wrote: hi, try exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1}) Note space before underscore. On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote

Re: [asterisk-users] Dropping call because extensions '200', 's' and 'i' doesn't exists

hi, try exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1}) Note space before underscore. On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote: this is my [from-trunk] extension: [from-trunk] exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1}) and this is [to-231] in sip_additional.conf:

Re: [asterisk-users] Setting a CDR field from using feature codes...

hi, you have not assign any value to CDR(userfield). try code = #111,self,SET(CDR(userfield)=111) On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez cur...@telecomabmex.comwrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule

Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?

hi, it is not difficult in php and mysql i have created a simple billing system for my wholesale postpay clients without any AGI. it report ACD ASR all calls ANSWERD calls filter by date by callerid etc. do billing as soon as call end. for billing i am using mysql trigger. report live calls. 2

Re: [asterisk-users] Setting a CDR field from using feature codes...

, Asghar Mohammad wrote: hi, you have not assign any value to CDR(userfield). try code = #111,self,SET(CDR(userfield)=111) On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez cur...@telecomabmex.com mailto:cur...@telecomabmex.com wrote: I am trying to set the CDR(userfield) to a certain

Re: [asterisk-users] Setting a CDR field from using feature codes...

/11/13 11:54 AM, Asghar Mohammad wrote: i am using exten = _XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)}) cli_name is field in mysql and it work fine. show me cli output without AGI. On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez cur...@telecomabmex.com mailto:cur...@telecomabmex.com

Re: [asterisk-users] Setting a CDR field from using feature codes...

OFICINA) in new stack -- Executing [h@oficina:2] NoOp(SIP/2003-000e, 2003) in new stack -- Executing [h@oficina:3] NoOp(SIP/2003-000e, ) in new stack On 4/11/13 12:24 PM, Asghar Mohammad wrote: how you are executing? show me your full context and how call enter in context

Re: [asterisk-users] sip set debug on output to file only (not to console)

hi, open debug only on problematic peer. sip set debug peer peer name or sip set debug ip peer ip On Fri, Mar 29, 2013 at 2:02 PM, Marie Fischer ma...@vtl.ee wrote: Hello everybody, I am trying to find an intermittent SIP error with one provider and thought the best first step would be to

Re: [asterisk-users] Need help about round-robin

Wimax and FH thanks and regards 2013/3/21 Asghar Mohammad asghar...@gmail.com hi, exten = _0612.,1,Set(CALLERID(number)=520460587) exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten = _0612.,n,Dial(Zap/r2

Re: [asterisk-users] Need help about round-robin

using a diguim cards 2013/3/22 Asghar Mohammad asghar...@gmail.com hi, i think we miss understood you Question? you need round robin on tdm trunk or on 2 internet connections? what are you asking about burden-sharing between Wimax and FH? On Fri, Mar 22, 2013 at 4:24 PM, Salaheddine

Re: [asterisk-users] Asterisk 1.8 and dual stack support

please see, http://lists.digium.com/pipermail/asterisk-users/2013-March/278130.html On Thu, Mar 21, 2013 at 5:47 PM, Jaap Winius jwin...@umrk.nl wrote: Hi folks, Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. As opposed to Asterisk 1.6.2.9 that I ran with squeeze,

Re: [asterisk-users] Need help about round-robin

hi, exten = _0612.,1,Set(CALLERID(number)=520460587) exten = _0612.,n,MixMonitor(zap_g2_${EXTEN}_${UNIQUEID}.wav|av(0}V(0)) exten = _0612.,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes) exten = _0612.,n,Dial(Zap/r2/${EXTEN},30,A(this-call-may-be-monitored-or-recorded) exten = _0612.,n,Hangup() Note r

Re: [asterisk-users] Howto create variable from the name of another one and get content of it

hi, ${myVar}STATUS is empty you have not assign any value here your var Set(__${myVar}STATUS=) is empty. use instead Set(__myVar=${ARG1}STATUS) and remove second line. On Thu, Mar 21, 2013 at 7:45 PM, Administrator TOOTAI ad...@tootai.netwrote: Hello, I have a variable created like ...

Re: [asterisk-users] Allow/Disallow

please post sip.conf. On Thu, Mar 21, 2013 at 8:01 PM, Nick Khamis sym...@gmail.com wrote: Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x8008000e

Re: [asterisk-users] Asterisk 1.8 and dual stack support

:) On Thu, Mar 21, 2013 at 10:27 PM, Jaap Winius jwin...@umrk.nl wrote: On Thu, 21 Mar 2013 12:21:44 -0700, Michael L. Young wrote: How are you determining that it is not listening on IPv4? bindaddr=:: should allow you to support dual stack. That's what I thought would happen. When I

Re: [asterisk-users] Delay before audio starts

hi, exten 000,1.Progress() work in some situation. On Thu, Mar 21, 2013 at 9:30 PM, Gerard gsara...@rarcoa.com wrote: On 03/21/13 14:14, Gerard wrote: I think a simple tcpdump of the traffic will show the mystery. It can be your provider doing something nasty. Have you tried using some

Re: [asterisk-users] Diagnosing call problem

hi Bharat, why you are giving same answer as mine over and over ? please read posts carefully. On Wed, Mar 20, 2013 at 6:48 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: Did u changed rtp.conf ? port is showing 39408. Asterisk definetly drop rtp packet for this port if not updated in

Re: [asterisk-users] AGI return codes

Hi ishfaq, if you want just loging some info into db you can do in dialplan without any AGI. i am doing billing on the fly in dialplan and mysql for every single user without AGI and enhanced call capacity almost double. let me know you need some examples. On Wed, Mar 20, 2013 at 12:56 PM, Ishfaq

Re: [asterisk-users] Diagnosing call problem

On Wed, Mar 20, 2013 at 7:11 PM, Asghar Mohammad asghar...@gmail.comwrote: hi, problem seem to client end i am going to install SFLPhone i will let you know when finish, have you check firewall on clients pc

Re: [asterisk-users] Diagnosing call problem

a successful call and a failed call, and I can see no difference except for things like port numbers and call IDs. It only fails occasionally, not on every call. Mitch On 03/20/2013 01:16 PM, Asghar Mohammad wrote: On Wed, Mar 20, 2013 at 7:11 PM, Asghar Mohammad asghar...@gmail.com

Re: [asterisk-users] Diagnosing call problem

hi, sflphone work fine installed and tested on debian with nat and without nat. please check setting in preferences my sflphone use alsa device. you should check with alsamixer maybe sometime mic get muted or you agent mute the mic. also check out what advice Mitch. NB. you can test with IAX also.

Re: [asterisk-users] Diagnosing call problem

hi satish, try to debug rtp on that ip and look rtp flow you can also test directmedia=no i encounter this as well i server is on public ip and clients connect via vpn , vpn server is also same asterisk server calls come in via public ip and go to call center via vpn i solved this by

Re: [asterisk-users] SIP account registration fails after upgrade to 1.8

hi, try srvlookup=yes On Tue, Mar 19, 2013 at 3:15 AM, Jaap Winius jwin...@umrk.nl wrote: Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the

Re: [asterisk-users] Diagnosing call problem

hi, rtp set debug ip 1.2.3.4 On Tue, Mar 19, 2013 at 2:09 PM, Mitch Claborn mitch...@claborn.net wrote: Thanks for the suggestions. 1) directmedia was taking the default of yes. I set to no. Will watch and see. 2) NAT is turned off (nat=no). I've never done any RTP debugging. Is that

Re: [asterisk-users] Diagnosing call problem

witch softphone you are using? on client pc installed some kind of virtualpc like vmware or virtualbox? client pc have more then one network interfaces? you can capture sip invites from soft phone by enabling debug on client ip sip set debug ip ip of softphon upload sip trace then somebody can

Re: [asterisk-users] Diagnosing call problem

rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad asghar...@gmail.comwrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy

Re: [asterisk-users] Need help understanding CDR

. In the given schematic what will be the Answered time and billed time. Thank you for the help in advance!! On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad asghar...@gmail.comwrote: If you have analog FXO ports then the call is considered answered as soon as dialing is completed

Re: [asterisk-users] Need help understanding CDR

distribution of time per call -- time in IVR, Queue, Call etc. Regards, Sans On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad asghar...@gmail.comwrote: hi, 00:00 -- Call Connected to asterisk - duration start here 00:01 -- welcome greeting starts billisec start here 00:11

Re: [asterisk-users] Need help understanding CDR

hi, billsec is time in seconds after call has answered, duration is total time in seconds of call. as your calls answered imidiatly your billsec and duration is almost same. On Sun, Mar 17, 2013 at 5:14 PM, RSCL Mumbai rscl.mum...@gmail.com wrote: Hi, Attached is a sample CDR. I need some

Re: [asterisk-users] Need help understanding CDR

If you have analog FXO ports then the call is considered answered as soon as dialing is completed not always true if FXO configured properly it should not send back answered as soon as dialed. On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling ewiel...@nyigc.com wrote: If you have analog FXO ports

Re: [asterisk-users] Sending SMS from asterisk

HI bilal, I don't think DAHDI can send SMS you have 2 options chan_mobile or chan_datacard ex chan_dongle chan_datacard i have not tested but with some mobile phones you can send sms i have tested also with some made in china unbranded phone that are capable to send and receive sms but not good

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