similar to: IAX2/0.0.29.199

Hey Guys, Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where? -Satish shirley*CLI> == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

Hey Guys, We upgrade asterisk from 1.2.x to 1.8.2.3 and my one of agi script doesn't working We have allpage.agi script for paging system on all polycom 501 but after upgrade it broke. Any idea what is this error ? extension.conf exten => 7770,1,agi(allpage.agi) exten => 7770,2,meetme(7770,dq) exten => 7770,3,playback(beep) exten => 7770,4,hangup following is agi debug....

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Following is my scenario to connect back to back PRI of two asterisk server. PRI cards are Sangoma A102D [Asterisk1]------------[PRI]-Cross Cable---------[Asterisk2] Asterisk1 ; Span 1 (MASTER) switchtype = national ; commonly referred to as NI2 context = from-pstn group = 0,24 echocancel = yes signalling = pri_net channel => 1-23 Asterisk2 ; Span 1 switchtype = national ; commonly

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

Dear ALL I want to transfer call from one phone 2 another phone so this is asterisk feature or SIP Phone feature or endpoint feature how can i transfer phone call from to another phone Rgd Satish patel --------------------------------- Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. -------------- next part -------------- An HTML

Hey guys, I have stdexten macro dialplan and I have to handle those who doesn't have voicemail box setup. Right now if someone call and if person unavailable the it's just hangup that call. I want it say "person doest have vm setup yet." smthing like that. How should I handle this in my dialplan ? -- Sent from my iPhone

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

Hey Guys, I knew this is stupid question but i just want to know what is the difference between Queue member logged out vs Pause ? -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110321/7646a66b/attachment.htm>

Hello Guys, I am trying to convert files that are .wac to mp3 after mixmonitor command is called but it doesnt execute the command, I tried the command in terminal it worked, any help please ... below is my dial plan exten=6500,n,Set(MIXMONITOR_EXEC=&& nice -n 19 /usr/local/bin/lame -b 8 -t -F -m m --bitwidth 8 --quiet "/var/spool/asterisk/monitor/${CALLFILENAME}.wav"

Hi All, I have the following section in my smb.conf file: [Satish' Hindi Alltime Hits - Gems] path = /mnt/d-drive/Music/Satish' Hindi Alltime Hits Gems browseable = yes public = yes readonly = yes guest ok = yes I am able to browse this share through win2k but not from winNT. But If I change the share

Hi, I am working of Intel 82576 device to test the SRIOV functionality. I want to assign VFs to a VMs(domU) and verify the control and data path. I installed Xen-unstable with 2.6.30-rc3(dom0). But the SRIOV functionality isn''t working, because MSI is not supported. Please let me know, what version of Xen and dom0 supports above functionality? Is there any tool available to manage SRIOV

Hello, I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device back to the SIP provider. Here is the setup: Some SIP phones are connected to an Asterisk System version 1.8. External connection to the public network is also done via SIP to a VoIP provider. Phone A has a CFW all calls to a phone number in public network (Mobile Phone) incoming call to

Hi: Suppose that I have a data frame as below x1 x2 x3 ... x10 wk1 wk2 ... Wk208 (these are the column names) For each record, x1, x2, x3 ... x10 are attributes. and wk1, wk2, ..., wk208 are the sales recoreded for this attribute combination. Suppose that now, that I want to do the following 1. Merge the data frame so that I have a new data frame grouped by values of x2 and x3 (for example).

I have RedHat 9.1 running. Installed Samba that comes along with the OS to test it out. When I was playing with it, I deleted the smb.conf file inadvertently. I uninstalled and reinstalled the Samba utilities, but that didn't help. How do I get back the default smb.conf file? Thanks in advance. Satish Kumar

Hey Users, I am using record application to record MeetMe conf. but look like its creating individual files for every channel. What applucation is best to record MeetMe conf ? ~ # ls -l /var/spool/asterisk/monitor/ total 489220 -rw-r--r-- 1 asterisk asterisk 44 Feb 16 08:42 8881-conf-20110216-084224.wav -rw-r--r-- 1 asterisk asterisk 1858284 Feb 16 13:05 8881-conf-20110216-130321.wav