search for: t1min

Hello! I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms): Teconisy most often doesn't answer the first invite before asterisk default t1min ended. Therefore asterisk sends one more invite. This second invite is answered by Teconisy with status 486 - Request terminated - Channel limit exceeded. (The second invite obviously is interpreted as...

...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 your network. No, that won't work. First ? 't1min', 'timert1' and 'timerb' are chan_sip, not pjsip options. Second ? the 'timer_t1' and 'timer_b' settings available in 'pjsip.conf' are global and not per-endpoi...

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

Hello List, I tried to change the following parameters in sip.conf file, but looks like it cannot be changed, Defaut values: ;t1min=100 ;timert1=500 ;timerb=32000 I have changed to: ;t1min=100 timert1=100 timerb=6400 Sometime I can see too many retransmission of BYE to some of the UAs if UA is unreachable. Is there a way that I can reduce the number of retransmission of BYE message? Regards Rajib -------------- next p...

...configure a remote user which happens to be listed in sip.conf (it's behind a NAT router but it registers with Asterisk, so... is it NATed or not?), and am surprised it actually rings and sends/receives voice with no problem, regardless of this parameter. I found discussions about using "t1min=500" in sip.conf, but it made no difference either. Has someone already experienced this and knows what can be done? Any hint much appreciated.

...= internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never promiscredir = no realm = asterisk recordhistory = no registerattempts = 0 registertimeout = 20 relaxdtmf = no sendrpid = no sipdebug = no t1min = 100 t38pt_udptl = no tos_audio = none tos_sip = none tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = ulaw,gsm subscribecontext = device-hints register => 6087294351:<sip password>@sip.broadvoice.com [trunk_1] type=peer...

...200 OK to Answer: exten => _X.,1,Ringing exten => _X.,n,Wait(1) exten => _X.,n,Answer exten => _X.,n,Goto(wherever) On further reading, I would think I could also solve it by setting the T1 values in sip.conf, instead of doing the above: Should I set "timert1=500", or "t1min=500" or both, or what? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

...ip signaling and it looks okay. i can provide it too if we required. here is my asterisk sip.conf kamailio context looks like [vmserver] type=friend context=default host=***local_ip_of_kamailio*** ; for below three i have tried all available options *directmedia=nonat directrtpsetup=yes nat=yes * t1min=500 disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm qualify=yes let me know how to solve this nating issue also i opened all required ports for sip. and rtp regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/piperma...

I am using 1.8. I need to change timerb to 6500, that is, if there is no response of some sort in 6.5 seconds, consider the call failed and try another route. It does not matter what do I set for the other timers: T1min=100 timert1=100 Timerb=6500 The command "sip show settings" always shows Timer B=32000. Any ideas how can I reduce Timer B?

...owtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = no checkmwi = 10 compactheaders = no defaultexpiry = 120 domain=sop-korniychuk domain=172.30.8.13 domain=172.30.8.13:5060 dumphistory = no externrefresh = 10 g726nonstandard = no notifyringing = yes srvlookup = yes t1min = 100 t38pt_udptl = no ;tos_audio = none ;tos_sip = none ;tos_video = none trustrpid = no useragent = Asterisk PBX usereqphone = no videosupport = no disallow = all allow = alaw type = friend host=dynamic context = noop-context dtmfmode=rfc2833 nat=no rtcachefriends=yes qualify=10000 deny=0.0.0.0/0...

...for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt: ?Hanging up call 70854efe-4157e3a8 at 10.168.7.103 - no reply to our critical packet (see doc/sip-retransmit.txt). I been googling this error and it was mentioned to use t1min= 500 however its only delaying the problem. ? any ideas on what is the cause of this problem. Only 2-3 atas are having this problem the rest are fine. ? Here is the sip debug the sip invites are not being received and in one of the message a busy response was sent back. ?Retransmitting #4 (no NAT)...

I have a concern with Dial command, I want to enable a secondary route with a remote partner, if the first route fails then we use the second one : Solution1: it will try both (there will be 2 simultanious actives calls ringing) this is not clean when calling an endusers exten => _X.,1,Dial(SIP/${EXTEN}@remote-sip1,5 <mailto:SIP/${EXTEN}@remote-sip1,5> ) exten =>

Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all