similar to: FWD and Asterisk

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

We all know the FWD is NO more available. How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite ? Any alternative for FWD ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm

I have 320 GB SATA HDD. When I checked my phpsysinfo, it shows 95% HDD is filled. [root at vicidialnow ~]# df Filesystem 1K-blocks Used Available Use% Mounted on /dev/sda2 301924504 285002780 1337472 100% / /dev/sda1 101086 11062 84805 12% /boot tmpfs 1553832 0 1553832 0% /dev/shm [root at vicidialnow ~]# du 16896 . You have new mail in /var/spool/mail/root [root at vicidialnow ~]# df -i

1) If I see the Loadavg more than 4 , whats the immediate solution to get it under 1 APART from restarting the server ? 2) I get too much of cross connections. Can Codec be the culprit ? I use g729. Can using GSM will solve the problem ? What could be the other reasons ? 3) Anyway to measure the bandwidth utilisation from the server ? -------------- next part -------------- An HTML attachment

We are using asterisk and sometime when our guys are on call , they hear some voice of person and amazingly that person is NOT from our center. Any one faced this kind of thing ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/7fe54bec/attachment.htm

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Which command to run which will auto detect all hardwares present in the system ? OS : CentOS Running Asterisk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090126/5e064cf8/attachment.htm

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

Looking for a Free VOIP Billing and Soft Switch. Any suggestions ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090211/9ff6e652/attachment.htm

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

I need to create 30 mins of GSM file for Asterisk . Silent / Blank file. Whats the best way to create it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100305/b92821c0/attachment.htm

In windows, we use BAT file to execute few series of command , which help us in not writing each command manually everytime we want to execute those commands. In CentOS, I want to do the same thing. Any Advice ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090125/d67fb239/attachment.htm

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Some at 5:34 pm EST DAILY, all my call get disconnect. I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its same. I tried changing VOIP provider I tried changing Internet Provider..But no help.. What could be the reason ? Here are my enties of crontab : ### recording mixing/compressing/ftping scripts 0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * *

Modified httf.conf file and added : ------------------------------------------------------ Alias /recordings/ "/var/spool/asterisk/monitorDONE/" <Directory "/var/spool/asterisk/monitorDONE"> Options Indexes MultiViews AllowOverride None Order allow,deny Allow from all </Directory> Created a folder under vicidial as recordings. FULL_RECORDING is also enabled.

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

What the best website and book to start learning asterisk ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090622/aabe17b8/attachment.htm

All my calls are getting DECLINED when I am trying from xlite : CLI shows : May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible: No pa th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256) May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/cc101-b790c1d8 compatible with