search for: timerb

Hi list, I have tried to set the value for "timerb" in sip.conf, general section and in user-context... tried on asterisk 1.4 up to version 13... no success. The value for timerb remains unchanged. (reload, restart, reboot.... all does not help...) sip show settings always show 32000ms for timerB. How can I configure the timerb value? thx,...

...e 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-endpoint. I cannot change...

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

On 10.12.2014 11:42, Frederic Van Espen wrote: > Hi, > > - Could you share the details of the SDP in each INVITE and OK packet? > - How are your SIP endpoints configured in asterisk sip.conf? (the SIP > trunk provider and the local endpoint) > - What type is the local endpoint? > > Cheers, > > Frederic > Frederic, I now have tried to describe the situation

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...

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 part -------------- An HTML...

.... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in between... I tried to "work around" this by increasing the settings for "timerb"... but I realized that asterisk does not care at all, what I set this value to... "sip show settings" always gives me 32000ms, and it does not make any difference if I configure timerb in the general context or in the phone context... any ideas? thanks, yves --- Diese E-Mail wu...

----- Original Message ----- > From: "Andrew Martin" <amartin at xes-inc.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Monday, May 11, 2015 1:35:07 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > > That should

Andrew Martin wrote: <snip> >> > Joshua, > > As a mitigation for this problem, could I increase the "timerb" option in sip.conf > to a large value, say 1 hour (instead of the default 32 seconds)? What other > consequences would there be from this change? I don't know if chan_sip will allow this, but if it does... it'll keep transmitting over and over... it would be better to get to t...

...t38pt_usertpsource: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL rtptimeout: NULL rtpholdtimeout: NULL sendrpid: NULL outboundproxy: PU.BL.IC.IP timert1: NULL timerb: NULL qualifyfreq: NULL constantssrc: NULL contactpermit: NULL contactdeny: NULL usereqphone: NULL textsupport: NULL faxdetect: NULL buggymwi: NULL auth: NULL fullname: NULL trunkname: NULL cid_number...

...hone > reports an error: > <0> | ERROR | receive a request with same cseq?? > > From the asterisk side, it never seems to receive this OK for CSeq 103, hence > the reason it sends out the INVITE again. > Joshua, As a mitigation for this problem, could I increase the "timerb" option in sip.conf to a large value, say 1 hour (instead of the default 32 seconds)? What other consequences would there be from this change? Thanks, Andrew

...5 5:42:57 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Andrew Martin wrote: > > <snip> > > >> > > Joshua, > > > > As a mitigation for this problem, could I increase the "timerb" option in > > sip.conf > > to a large value, say 1 hour (instead of the default 32 seconds)? What > > other > > consequences would there be from this change? > > I don't know if chan_sip will allow this, but if it does... it'll keep > transmitting ove...

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 =>