search for: allowoverlap

...11201's ATA (SPA2102). Shouldn't the UDPTL stream go through Asterisk? Have i missed sometheng else? Asterisk SVN-trunk-r247652M built by root @ sip on a i686 running Linux on 2010-01-25 11:10:15 UTC [0197673581] secret=xyz callerid=Input Interior Orebro (fax) disallow=all allow=alaw:40 allowoverlap=yes allowsubscribe=yes callcounter=yes callingpres=allowed_passed_screen canreinvite=no context=inputinterior.se directmedia=no dtmfmode=rfc2833 faxdetect=no host=dynamic language=se nat=yes qualify=yes sendrpid=pai t38pt_udptl=no transport=udp trustrpid=yes type=friend videosupport=no [0851711201...

Hello I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no rtcachefriends = yes externaddr = my ip address externhost = my domain address ;   foo.dyndns.net; refreshed periodically externrefresh = 180       localnet = 172.31.40....

...access the server (bots) and they have been able to make calls (in today's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid="blah" <XXXXXXXXXX> host=dynamic nat=yes canreinvite=no...

...17036361355,2,SayDigits(${EXTEN}) exten => 17036361355,3,Goto(testdtmf|s|1 ;Ring on Elle mobile phone. ;exten => s,1,Answer() ;exten => s,n,Dial(SIP/17037171234,150,r,t,) [general] register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146 registertimeout=20 context=incoming allowoverlap=no bindport=5060 bindaddr=192.168.1.10 srvlookup=no ;context=incoming ; The SIP provider [voipvoip.com] canreinvite=no username=1112530146 fromuser=1112530146 secret=albany!@#123 context=incoming type=friend fromdomain=sip3 at voipvoip.com host=69.90.209.57 dtmfmode=rfc2833 disallow=all allow=alaw...

...Any help rendered by you in this subject is greatly appreciated. I have been breaking my head trying to resolve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writep...

On September 8, the Asterisk development team released the AST-2016-007 security advisory. The security advisory involved an RTP resource exhaustion that could be targeted due to a flaw in the "allowoverlap" option of chan_sip. Due to new information presented to us by Walter Doekes, we have made the following updates to the advisory. In the "Description" section, the following text has been added: UPDATE (20 October, 2016):...

....2.1 on an ubuntu on AWS, which is > experiencing a > drop in call. It does not have a certain time, it is random. The > audio > is flowing normally and the call is dropped. > Has anyone ever experienced this? > > My settings changed below: > > allowoverlap = no > udpbindaddr = 0.0.0.0 > tcpenable = no > tcpbindaddr = 0.0.0.0 > > transport = udp, ws, wss > > srvlookup = yes > > directmedia = no > > rtcachefriends = yes > > externaddr = my ip address > > externhost = my do...

Hi all, I want to two sip clients connect through Asterisk in local network for testing. My sip.conf file looks like this [general] context=internal allowguest=no allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=no disallow=all allow=ulaw alwaysauthreject=yes canreinvite=no nat=yes session-timers=refuse localnet=192.168.1.0/255.255.255.0 [7001] type=friend host=dynamic secret=123abcd context=internal [7002] type=friend host=dynamic secret=456abcd context=intern...

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97 disallow=all allow=ulaw allow=ilbc mohinterpret=default mohsuggest=default language=en useragent=TCTC PBX ;dtmfmode = info fromdomain=10.10.60.253 ;relaxdtmf=yes [15]...

Hi, I'm trying to get a softphone configured. In Sip.conf [general] I found an example that said I need: nat=yes localnet=192.168.xxx.xxx Is localnet supposed to be a LAN IP or a WAN IP? Thank you, Gary

...t 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in advance Regards Bie below is my sip.conf allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm I also had 2 extensions (me at 250 and 998 is my SPA-3102) and my users.conf goes below: [general] fullname=New User userbase=6000 hasvoicemail=yes vmsecret=1234 hassip=yes hasiax=yes hasmanager=no...

...the call to be disconnected after! By checking the logs, I can see thischan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission Mjk3MGU1NjgxZWQwM2E3MjhjZmFiNzhjOGVjZjg5ZTc for seqno 2 (Critical Response) Here's my simple sip configuration[general]context=internalallowguest=noallowoverlap=nobindport=5060bindaddr=0.0.0.0srvlookup=nodisallow=allallow=ulawalwaysauthreject=yescanreinvite=nonat=yessession-timers=refuseexternip=<IP>[7001]type=friendhost=dynamicsecret=123context=internal[7002]type=friendhost=dynamicsecret=456context=internal A snoop capture for my call is uploaded i...

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx...

...ecallerid=yes callerid=asreceived cidstart=ring hidecallerid=no immediate=no pickupgroup=1 ;context=incoming channel => 1-4 Sip.conf [general] srvlookup=yes ;allows DNS lookups of server names naxexpirey=180 defaultexpirey=160 context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) tos_sip=cs3 tos_audio=ef ; bindport is the local UDP port that Asterisk will ; listen on bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) srvlookup=y...

...lp anyone assist me in this problem - let me know if I missed something. It *feels* like an Asterisk bug but maybe a SIP expert can spot the problem in signalling/RFC conformance.. Thanks in advance, Chris Bennett -------------- next part -------------- [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes domain=proxy.myhostname disallow=all allow=alaw sipdebug = yes recordhistory=yes dumphistory=...

...User Phone 192.168.97.74/24 There is a Lan2Lan VPN tunnel between the Firewall2 and the Remote Office Firewall3 I can Ping the remote office phone from the asterisk PBX at all times. Now I have my Sip.conf setup with externip= X.Y.Z.250 [general] port = 5060 bindaddr = 0.0.0.0 context = default allowoverlap=no srvlookup = yes : externip = externip = x.y.z.250 localnet=10.202.17.0/255.255.255.0 qualify=yes nat=yes register = xxxxxxx:SipServer/xxxxxxxx limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes useclientcode=yes canreinvite=no I have pfsense setup to forward ports 5060 and RTP...

...,Noop(From is ${SIP_HEADER(From)}) exten => xxx,n,Set(CALLERID(num)=1234567890) exten => xxx,n,Set(CALLERID(name)=) exten => xxx,n,Noop(CallerID is ${CALLERID(all)}) exten => xxx,n(dialout),Dial(SIP/POTS1,60,o) exten => xxx,n,Hangup And my general and section from sip.conf [general] allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw limitonpeers=yes notifyringing=yes maxexpirery=180 defaultexpirey=180 [POTS1] type=friend secret=xxx context=pots_in host=dynamic dtmfmode=info disallow=all allow=ulaw allow=alaw canreinvite=no...

...authentication issue going out over the same sip account. It appears that my server isn't sending the second invite after proxy authentication request. I can't figure out why; any ideas would be greatly appreciated. Thanks! - Josh Here is my sip.conf: [general] context = default allowoverlap = no bindport = 5060 bindaddr = 0.0.0.0 srvlookup = yes externip = 172.21.235.2 localnet = 172.21.235.2/255.255.0.0 dtmfmode = rfc2833 relaxdtmf = yes disallow = all allow = ulaw allow = gsm maxexpirey = 30 defaultexpirey = 180 relaxdtmf=yes canreinvite = no nat = 0 UserAgent = Asterisk echocancel...

...s.conf: --------- [general] [externe] exten => 555,1,Dial(IAX2/111) exten => 555,n,Hangup() [special] exten => 111,1,Dial(IAX2/111) exten => 111,n,Hangup() [default] exten => 444,1,Dial(IAX2/444) exten => 444,n,Hangup() - Sip.conf (SIP server): [general] context=default allowoverlap=no udpbindaddr=0.0.0.0 tcpenable=no tcpbindaddr=0.0.0.0 srvlookup=yes --------- - Logs server: --------- -- Accepting AUTHENTICATED call from 10.0.100.238: > requested format = gsm, > requested prefs = (), > actual format = ulaw, > host prefs = (),...

...21@default : State:Unavailable Watchers 4 --------------- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind...