search for: delayreject

...r connected directly to the internet and the other behind a NAT. ---- The iax.conf file for the one that is directly connected to the internet is as follows: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=yes mailboxdetail=yes #include iax_additional.conf [I'm using AMP as the GUI interface]...

...otraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=>1 -------------------------------------------------- (same for SERVER2) IAX.conf ------------------------------------------------ ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register => username:password@server2.domain.com ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montr?al ~ host=Dynamic ~ secret = password...

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255 1207 OK (33 ms) Why is it running on port 1207?

...low. Anyone know why even when SIP has registered I do not see incoming calls? Thanks, ------extensions.conf---------------- [global] [general] bindport=5060 bindaddr = 0.0.0.0 deny=0.0.0.0/0.0.0.0 externhost=xyz.dyndns.org localnet = 192.168.1.0/255.255.255.0 disallow=all allow=ulaw allow=gsm delayreject=yes nochecksums=no allowguest=no delayreject=yes pedantic=no register => 703XXXYYYY at sip.broadvoice.com:s ecurepassword:703XXXYYYY at sip.broadvoice.com/5000 [5000] type=friend context=internal-phones secret=xxx qualify=yes host=dynamic ; behind nat dtmfmode=rfc2833 [5002] type=friend conte...

Hello, I have just set up an asterisk box (Debian unstable) and I would like to test it with a H.323 application (gnomemeeting). When I call the demo voice menu, I can't hear any sound. asterisk says that the soundfile is played: -- Executing BackGround("H323/ip$212.9.189.172:30005/29597", "demo-congrats") in new stack -- Playing 'demo-congrats' (language

...ested at this time (1:0/0/1)". So, i don't know where is the probleme : is it from iax accounts loaded from the database, or the new account 444 ??? Below are the conf files and verbose output. Thank you very much for your help :) --------- - iax.conf --------- [general] bindport=4569 delayreject=yes language=fr autokill = yes calltokenoptional = 0.0.0.0/0.0.0.0 minregexpire = 60 maxregexpire = 500 mohsuggest=default careinvite=no rtcachefriends=yes [444] type=friend host=dynamic context=special secret=iop --------- - extconfig.conf: --------- [general] [settings] iaxusers => mysql,...

...ault is to bind to all ; local addresses). ; [general] bindport=4569 ;bindaddr=192.168.0.1 ; ; Set iaxcompat to yes if you plan to use layered ; switches. It incurs a small performance hit to enable it ; ;iaxcompat=yes ; ; For increased security against brute force password attacks ; enable "delayreject" which will delay the sending of authentication ; reject for REGREQ or AUTHREP if there is a password. ; delayreject=yes ; ; You may specify a global default AMA flag for iaxtel calls. It must be ; one of 'default', 'omit', 'billing', or 'documentation'. The...

How do I get Gnophone to register to my Asterisk server? I have set up iax. conf as follows: [tim] type=friend ;username=tstornes host=dynamic ;defaultip=207.194.60.56 secret=1111 context=from-iax callerid => "Tim" <5000> auth=plaintext qualify=10 permit=0.0.0.0/0.0.0.0 and extensions.conf includes a section in the context from-iax: exten => 5000,1,Dial(IAX/tim/s|100|r)

...ll addresses on machine) disallow=all allow=ulaw allow=alaw allow=gsm context = from-sip ; Send unknown SIP callers to this context And in iax.conf (snippet) : [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) ;delayreject=yes disallow=all allow=ulaw allow=alaw allow=gsm jitterbuffer=yes mailboxdetail=yes authdebug=yes

...he server about 2 months ago. After eliminating potential hardware problems, etc., I figured out that when I change jitterbuffer=yes to jitterbuffer=no in iax.conf the audio delay problem is solved, but the call quality degrades. My prior settings were: [general] bindport = 4569 bindaddr = 0.0.0.0 delayreject=yes disallow=all allow=ulaw allow=alaw mailboxdetail=yes register => XXXXX@voip.teliax.com dropcount=3 jitterbuffer=yes maxjitterbuffer=500 maxexcessbuffer=300 canreinvite=no Any ideas on tweaking iax.conf to optimize call quality, but avoid the audio delay with forwarded calls? Regards, Mik...

...92966% 99.999321% <br> --- Results after 14 passes --- <br> Best: 99.999 -- Worst: 99.993 -- Average: 99.996930, Difference: 99.997334 <br> <br> ; iax.conf server_master <br> ; <br> [general] <br> bindport=XXXX <br> bindaddr=222.222.222.222 <br> delayreject=yes <br> language=fr <br> allow=alaw <br> maxjitterbuffer=800 <br> trunktimestamps=yes <br> tos=ef <br> <br> [jourdain] <br> type=friend <br> host=111.111.111.111 <br> port=XXXX <br> context=iax-jourdain <br> trunk=yes <br&g...

...channel(s) Best regards, Alex Massover CTO Gamcom Ltd. P.S. Our company have ordered your hardware cards. -------------- next part -------------- [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) tos=0x14 delayreject=yes disallow=all allow=g273.1 allow=g279 allow=ilbc allow=ulaw allow=alaw allow=gsm allow=g726 jitterbuffer=no mailboxdetail=yes notransfer=no ;register => gamcom:2046590@iaxtel.com ;register => 478933:2046590@iax2.fwdnet.net ;[fwd-gw] ;type=peer ;auth=md5 ;secret=2046590 ;username=478933 ;...

...h State 66.234.228.170:4569 XXXXXXXXX 208.184.214.241:4569 60 Registered ----------------------- My config is: iax.conf:[general] iax.conf:disallow=all iax.conf:allow=ulaw iax.conf:allow=ilbc iax.conf:allow=gsm iax.conf:allow=adpcm iax.conf:allow=alaw iax.conf:jitterbuffer=no iax.conf:delayreject=no iax.conf:register => XXXXXXXX:XXXXXXX@gwiax-in-01.voicepulse.com iax.conf: iax.conf:[voicepulse-in-01] iax.conf:type=user iax.conf:context=voicepulse-test iax.conf:auth=rsa iax.conf:inkeys=voicepulse01 extensions.conf:[general] extensions.conf:static=yes extensions.conf:writeprotect=yes exte...

...------8<-------8<------8<------8<------8<------ Config files (colo* and office* have identical configs except where noted): iax.conf: [general] disallow=all allow=ulaw allow=gsm tos=0x18 pingtime=1 lagrqtime=1 ; tried iaxcompat=no and yes, no apparent difference iaxcompat=yes ;delayreject=yes jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=100 minexcessbuffer=25 jittershrinkrate=1 (colo* has an entry for office-ast, office* has an entry for colo-ast with the host entries set respectively) [office-ast] type=peer host=192.168.2.1 qualify=500 trunk=yes [phone] type...

...efined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!! I'm going crazy... Here's my config: iax.conf: [general] bindport=4569 bindaddr=dynamic nochecksums=yes delayreject=yes language=it bandwidth=high disallow=all allow=alaw jitterbuffer=no forcejitterbuffer=no minregexpire=300 maxregexpire=600 tos=lowdelay autokill=yes [support] type=friend auth=plaintext,md5 username=support secret=support host=dynamic qualify=no context=incoming-iax extensions.conf (excerpt):...

Hello, I'm looking for some advice on securing Asterisk. Recently my servers been under several brute-force SIP attacks. I have several remote sites, as well as many roaming users, who may have PC softclients and/or SIP based hardphones. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

..._here bindaddr=to_this_address srvlookup=yes maxexpiry=7200 defaultexpiry=3600 [authentication] [test-voice] type=friend host=dynamic secret=not_my_secret context=users disallow=all allow=ulaw nat=yes directmedia=no qualify=yes trunk=no IAX2 - [general] bindport=bind_here bindaddr=to_this_address delayreject=yes disallow=all allow=g722 allow=ulaw allow=gsm jitterbuffer=no encryption=yes [test-fax1] type=friend host=dynamic username=test-fax1 secret=not_my_secret context=users disallow=all allow=ulaw qualify=yes trunk=no requirecalltoken=no SIP peers are Zhone ZNID-2xxx series ONTs. IAX peers are AT...