similar to: Timer B in sip.conf cannot be changed

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?

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 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, yves --- Diese E-Mail wurde von Avast

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

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 =>

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

In article <mfbt6f$9rt$1 at softins.softins.co.uk>, Tony Mountifield <tony at softins.co.uk> wrote: > I am trying to debug a SIP issue, between an Asterisk 1.2.32 system that > is behind a network device to which I don't have ready access, which is > performing NAT with possibly some kind of SIP ALG, and an Asterisk 11 > system on a public IP. > > My question is

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... 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

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

Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Asterisk Project Security Advisory - AST-2016-002 Product Asterisk Summary File descriptor exhaustion in chan_sip Nature of Advisory Denial of Service Susceptibility Remote Unauthenticated Sessions Severity Minor

? HI ? I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn) ? Out going calls from asterisk to the ata works fine Incoming calls from the ata to asterisk cuts off with the error msg ? Maximum retries exceeded on transmission 70854efe-4157e3a8 at 10.168.7.103 for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Dec? 6 13:52:43] WARNING[3921]: chan_sip.c:3858

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

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

Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. Thanks, Josh Fuller josh.fuller at

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

Dear all, I have a question about session timers. I have one of my installations (* 1.6.2.7) where all SIP calls get stuck, like this: cs4wall*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry 192.168.40.178 42 3c291b87c66e-sl 0x8 (alaw) No Rx: BYE 192.168.40.179 41

Hello, I have my sip stuff seemingly working fine as well as my zaptel stuff working great... But I have a problem with sip registration timers (I'm guessing here). In my extensions.conf file I have this... exten => 2244,1,Dial,Zap/2|25 exten => 2244,2,Dial,Sip/brian|25 exten => 2244,3,VoiceMail,u2244 But if I close my sip phone and a call goes through it will still wait the 25

Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags :