similar to: Howto see the source ip address of SIP call in cli monitor

Hi, Would like to run the software to monitor the quality of the bandwidth. Suggestions welcome? Thank you. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081211/85bd0069/attachment.htm

Hi, There is a long call setup time untill the call connects. How can I play a beep tone say every 4 seconds to the caller untill the call connects? Tx. Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090304/38e17d3e/attachment.htm

Please help... The 1st voicemail message after a reload has audio to the caller. All subsequent calls have no audio to the caller even though the same voicemail application is being called? Asterisk Version 1.4.21.2 Executing [0872200189 at In:2] VoiceMail("SIP/voip-1fd034e0", "910|u") in new stack -- <SIP/voip-1fd034e0> Playing 'vm-theperson' (language

Say, I'm looking for a simple way to dial a number repeatedly for two minutes at a time. The purpose is to busy up a faulty analogue line in an incoming hunt group. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100410/0d4e92e9/attachment.htm

Hi, I've tried the link http://www.asteriskguru.com/tools/audio_conversion.php but it returns an error at the moment. Any other ideas most welcome. Tx Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090423/c491a7b9/attachment.htm

Hi, incoming SIP calls have a channel name in the form of: SIP/<ip-adresss-of-peer>-<handle> This is a way to get fetch the IP address of the remote side of a SIP call - in most cases. However, sometimes, instead of the IP address, there is a host name in the channel name. I assume, this value in the channel name is not the real IP address, but just a field filled in by the remote

As per my problem yesterday with the Cisco 7960 and getting it talking to Asterisk on a different subnet, I gave up trying and just put the Asterisk box back on the internal subnet. However, I made two changes: - the external IP address is set on an ethernet alias eth0:0 - the main Linux router will change outgoing requests from 10.1.1.2 to the external IP (rather than the default behaviour of

Hi, Why does this warning occur and what are the implications of it? I'm concerned about calls never getting hung up.....! chan_sip.c:12890 handle_response: Remote host can't match request CANCEL to call '2f197e56611061a678c13b881b2691a9 at 411.2.139.106'. Giving up. Tx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Please help. Asterisk 1: Sip.conf [VoipDirect777821] type=friend host=dfvvd.dyndns.org username=VoipDirect777821 secret=xxxxxxxxxxxx accountcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12

Can I use grep ? Tried but not working. please help Thanks Shaun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081212/c856a1f1/attachment.htm

Say, I just picked this up on my messages! There are a whole host of these requests! Anyone know whow there people are? Is there a way to report them? Any suggestions as to how to block them? [Aug 23 10:34:16] NOTICE[1010] chan_sip.c: Registration from '"912" <sip:1 at 41.1.1.1>' failed for '184.106.217.112' - Wrong password [Aug 23 10:34:16] NOTICE[1010]

Does anyone have a recommendation for an Open Source IP Address Management solution please? There are several commercial players such as BlueCat, BT Diamond, InfoBlox, VitalQIP. And, Solarwinds makes a module that focuses on IPAM. Most vendors tie logic into DNS and DHCP into IPAM designs. In any case, does anyone have awareness of an Open Source solution? Thank you Tom -------------- next part

Does anyone know how to change the source IP address/Source Interface of RTP packets? Changing the SIP source IP address in sip.conf has no apparent impact on RTP. RTP traffic still uses the address assigned to the outbound interface.

Hi all! I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP addresses from the same subnet set on one interface, and bindaddr is set to the second on them in sip.conf and in iax.conf. Incoming connections work as expected. However, for outgoing connections it seems that asterisk tells the kernel to use the specific "bind" address only in case of UDP usage (both SIP and

Hello, how do you block incoming AND outgoing traffic to a site? I have 2 drop lines for a site in my /etc/sysconfig/iptables: *filter :INPUT DROP [0:0] :FORWARD DROP [0:0] :OUTPUT ACCEPT [294:35064] -A INPUT -m state --state RELATED,ESTABLISHED -j ACCEPT -A INPUT -i lo -j ACCEPT -A INPUT -s xx.xx.xx.0/24 -j DROP -A INPUT -d xx.xx.xx.0/24 -j DROP -A INPUT -p icmp -m icmp --icmp-type any -j

On Sat, Mar 11, 2017, at 11:50 AM, Kseniya Blashchuk wrote: > Hey guys, any thoughts on that? Probably a bug or is it a default > behavior? I'd suggest providing the configuration to make sure it is correct. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org

Hey everyone -- Is there a way to refer to the server with the tftpboot images so that when a client is booting diskless, it can mount an nfs directory to that tftp server? That is to say that I have a tftp server on 192.168.0.244. The usr directory is set to be shared amoung the diskless clients via nfs. In the root image, I have /etc/fstab set to something like: blah blah

Say anyone know howto debug this: Failed to initialize G.729 copy protection! X64 CentOS system. Running Asterisk as Non-root. Downloaded latest G729 driver and registered it sucessfully. Restarted Asterisk. But still get this error! Asterisk 1.4.21.2 built by shaunw @ xxx.xxx.biz on a x86_64 running Linux on 2008-08-06 19:11:02 UTC Intel(R) Xeon(R) CPU E5420 @ 2.50GHz show g729 No

Hi Can you please help me make this into one statement... It doesn't work if I say _9000[1-9]0[1-8]. Also would like to be able to achieve _9000[1-9]0[1-8]XXXXXXXX, Asterisk 1.4 exten => _900010[0-8].,1,Goto(route1,${EXTEN:5},1) exten => _900010[0-8].,2,Hangup exten => _900020[0-8].,1,Goto(route,${EXTEN:5},1) exten => _900020[0-8].,2,Hangup exten =>

On Mon, Mar 13, 2017, at 03:52 AM, Kseniya Blashchuk wrote: > Hi! > Attached sip.conf and interface config as well. In this case we use only > TLS, but I have checked with TCP - same situation, 192.168.0.172 is used > as > a source. For UDP 192.168.0.177 is used as expected. Does the output of netstat -a confirm that it is bound to only that IP address? If so, then it seems