search for: compactheaders

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2010 Oct 12
0
rtpip patch
...le I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@ #define DEFAULT_CALLCOUNTER FALSE #define DEFAULT_SRVLOOKUP TRUE /*!< Recommended setting is ON */ #define DEFAULT_COMPACTHEADERS FALSE /*!< Send compact (one-character) SIP headers. Default off */ - -#define DEFAULT_RTPIP "auto" - #define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */ #defi...
2010 Nov 03
1
inbound call issue...
...241-538634340-1288768121281- Max-Forwards: 70 Content-Length: 0 Here's the configs: subscribecontext = device-hints allowexternaldomains = yes allowguest = yes allowsubscribe = yes allowtransfer = yes alwaysauthreject = no autodomain = no callevents = no canreinvite = yes checkmwi = 10 compactheaders = no defaultexpiry = 120 dumphistory = no externip = 216.26.109.22 g726nonstandard = no jbenable = yes jbforce = no jblog = no localnet = internal subnet maxcallbitrate = 384 maxexpiry = 3600 minexpiry = 60 mohinterpret = default nat = yes notifyringing = yes pedantic = no progressinband = never pr...
2007 Sep 18
1
stanaphone issues. can someone verify my config?
...s bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=no allow=all allow=g729 rtptimeout=4 (tried this on the default of 30 and it just makes it take longer to give the error, and I like it low incase the internet dies I don't end up talking to nothing for a long time without realizing it.) compactheaders = yes externip = 60.xxxxxx (our static IP is here) localnet=192.168.0.0/255.255.0.0; nat=yes canreinvite=no ; richards stanaphone incoming to ext 8800 register => 089xyz:xxxxxxxx at sip.stanaphone.com/8800 ; richards italk to ext 8800 register => 64997xxxxx:xxxxx at akl.italk.co.nz/8800...
2007 Oct 23
1
Short format of SIP INVITE - how to change
My Asterisk box send INVITEs in the short form, i.e., "f:" instead of "from", "v:" instead of "via" and so on. Is there a way to force asterisk to use full format? thanks Vitaly __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
2011 Oct 24
0
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
...esult. What I did not configured? My sip.conf [general] context = default allowguest = no bindport = 5060 bindaddr = 0.0.0.0 allowexternaldomains = no allowoverlap = yes allowsubscribe = yes allowtransfer = 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...
2006 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
...allow=g729 allow=gsm allow=ulaw jitterbuffer=yes maxjitterbuffer=1500 ;allow=ilbc ;musicclass=default ;language=en ;relaxdtmf=yes rtptimeout=60 ;rtpholdtimeout=300 ;trustrpid = no ;sendrpid = yes ;progressinband=never ;useragent=Asterisk PBX ;promiscredir = no ;usereqphone = no dtmfmode = rfc2833 ;compactheaders = yes ;sipdebug = yes ;subscribecontext = default ;notifyringing = yes And these are the extensions: [xxxx] type=friend host=dynamic dtmfmode=rfc2833 username=xxxx secret=xxxx [xxxx2] type=friend host=dynamic dtmfmode=rfc2833 username=xxxx secret=xxxx As you can see I put the jitterbuf...
2011 Jul 19
3
CentOS 6
I finally switched workstations, and am running into a lot of truly annoying details with CentOS 6, and the software with it. For one, I'm doing this by webmail, and this version of Firefox *insists* on putting what I'm typing in gray, rather than black. For another, I despise the new version of thunderbird, since it now shows the full subject, if I have what they used to call the preview
2009 Aug 04
0
SIP server behind NAT
...agent string > promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address > ;usereqphone = no ; If yes, ";user=phone" is added to uri that contains > dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 > ;compactheaders = yes ; send compact sip headers. > ;sipdebug = yes ; Turn on SIP debugging by default, from > ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests > ;notifyringing = yes ; Notify subscriptions on RINGING state > ;alwaysaut...
2006 Dec 18
0
pap2/wrt54gs/asterisk
...ult: rfc2833 ; Other options: ; info : SIP INFO messages ; inband : Inband audio (requires 64 kbit codec -alaw, ulaw) ; auto : Use rfc2833 if offered, inband otherwise ;compactheaders = yes ; send compact sip headers. ;sipdebug = yes ; Turn on SIP debugging by default, from ; the moment the channel loads this configuration ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests...
2011 Feb 10
2
Unable to make outgoing calls with Internode
...one number> ;fromdomain = internode.on.net ;secret = <secret> ;qualify = yes canreinvite = no ;auth = <phone number>:<secret>@BroadWorks ;nat = never ;pedantic = yes ;insecure = invite,port ;ignoresdpversion = yes ;compactheaders = yes As you can see I've tried lots of settings. It registers and peers with the provider, but no outgoing. The provider can call me though. In extensions.conf: [internode-outgoing] exten => _X.,1,Dial(SIP/${EXTEN}@sip-out) exten => _X.,n,Answer(2) exten...
2005 Jan 05
5
Asterisk with MySQL
You are reading the instructions for the STABLE 1.0 version of asterisk and are using the CVS version. Goto the wiki and read the instructions for RealTime. -Matthew ----- Original Message ----- From: "Muhammad Rizwan Khan" <rizwan@advcomm.net> To: <Asterisk-Dev@lists.digium.com> Sent: Wednesday, January 05, 2005 12:42 PM Subject: [Asterisk-Dev] Asterisk with MySQL >