similar to: Sip, Qualify=200 that doesn't qualify. How to signal this state to the Peer

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 guys, I configure one Fedora Core Linux 5 for use with asterisk as gateway using Digium TE110P interconected in Alcantel 4100 I've set up it to register 100 voip numbers on my provider. All calls on Alcatel is send to asterisk. In some periods of day i receive this messages on asterisk console: Dec 12 17:49:30 NOTICE[11565]: chan_sip.c:11564 sip_poke_noanswer: Peer

Although I'm convinced that Broadvoice doesn't have the most stable of ping times, it seems like I get ping results that are approximately the ping time +2000ms at times. Has anyone experienced this problem with qualify on a SIP connection before? So here, was the ping 20ms or 2020ms as reported? Aug 22 06:39:49 NOTICE[6964]: chan_sip.c:8481 handle_response_peerpoke: Peer

I would like to ask, if someone observe also problem with peer qualify problems, my asterisk log is full with UNREACHABLE/REACHABLE messages, even when two asterisks are in LAN environment, please take a look into this debug, I can't find any problem with packet loss, all qualify requests are replied and acknowledged, I will submit bug report, if you will also not find any problems here...

Hi there, could anybody tell me if the info below is still correct: Note: In order to obtain useful DIALSTATUS information when dialing a peer you will need to have qualify=yes in that peer's definition (e.g. in sip.conf or iax.conf). http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS THANKS!! -- Regards, Rustam Kovhaev

Hi, I think I encountered a bug in the qualify timers for IAX on asterisk 1.8 but I'd like to check if I'm not messing up in my config somewhere before reporting a bug. In my IAX peer configuration I have this: [remote-host] type=friend host=172.16.6.45 username=remote-host secret=test notransfer=yes qualify=16000 qualifyfreqnotok=30000 disallow=all allow=alaw allow=ulaw allow=ilbc

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:

On 2016-08-09 10:06, Faheem Muhammad wrote: > trip time and Call Setup time of SIP Requests. > In case of GSM Network with high delay you need to set the T1 timer a > higher value like 1000ms (500 ms default). Similarly you can reduce the > Call setup time by configuring 'T2' upto you choice as per you telephony > network. Configure t1min, timert1 and timerb according to

Hi, We have been migrating our PBX system from Asterisk 1.8 and chan_sip to Asterisk 13 and chan_pjsip. Things are mostly, ok, but now I have stumbled on a behaviour difference I don't like. With chan_pjsip when a phone went unexpectedly offline (Ethernet cable disconnected) Asterisk would detect this quickly (through the 'qualify' pings), mark the phone as 'Unavailable' and

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

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

Using zoiper on a nexus 4, asterisk 11.5.1, sometimes we see failed authentication. The secret seems correct, so we can't figure out why we're getting failed authentication. But at the same time the device shows as registered: [Oct 8 18:14:14] NOTICE[510]: chan_iax2.c:11071 socket_process_helper: Peer 'n4' is now REACHABLE! Time: 441 [Oct 8 18:15:58] NOTICE[519]:

Hi All, I'm wondering if anyone can share any info on why I frequently get peer timeouts like below, and receive 489 messages from another A*k server on the same LAN. For the peers, we've one L2 switch. ICMP is <1ms. The CPU of the main A*k server is usually < 2%. So I can't see why we'd get such large delays. The phones are all Cisco 7940s (SIP 2xx) The 489 originate

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 =>

Hi, Has anyone had experience installing it? yum install asterisk-chan_misdn I'ts the latest Trixbox Distro version and same issues exists if add in the Trixbox repo. FAILS as per below: I have a ISDN single port PCI BRI card installed and detected. __________________ Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * addons: www.ftp.saix.net * base:

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