similar to: @LongLink

. Monday, July 5, 2004 15:50:06 (-08:00hrs UTC) Hello asterisk-users, From the following post: On Mon, 5 Jul 2004, Joe Baptista wrote: > On Mon, 5 Jul 2004, Randy Bush wrote: >> i did not criticize the protocol. remember, my question started >> with >> >> >> i am looking at iax to

>> auth: in openpam_parse_chain(): /etc/pam.d/dovecot(1): missing or invalid facility > > I do not think that it has something to do with the dovecot settings > itself but perhaps with the pam facility settings instead? i can believe that. any clues to debug? randy

i try to place a call exten => _X.,1,Dial(SIP/${EXTEN}@foo:5061,60,Ttr) where sip.conf has an entry [foo] secret=torture callerid="local ext 103" <19146665555> type=friend fromuser=asterisk auth=both host=dynamic canreinvite=yes context=in-914 mailbox=001 i get May 22 23:11:31 WARNING[140400128]: chan_sip.c:902 create_addr: \

i have extensions in locations across a number of telco area codes. when someone in seattle picks up and dials 91234567, it would be nice to transform it to 92061234567. i would prefer not to have an extension context per area code. it would be cool to be able to set a variable in the sip.conf bit for each phone with it's geographic default area code. or other folk may have a better hack.

if i go off hook and dial 666 from an internal sipura spa-x000 (at extn 141), it rings straight through to extn 666. using the same dialplan, from a cisco 7960 with 7.1 sip code (at extn 142), i have to go off hook hit NewCall punch 142 (or any valid extn in the dialplan) hit Dial then dial 666 wtf? sip.conf for crisco [fiji] callerid="crisco" <142>

i am looking at iax to see if it is applicable to my needs. i would appreciate any corrections of what i think i have understood but probably have not. iax uses udp and traverses nats. neither of these seems useful to me. i loathe nats, and udp is not well-behaved in the sense of congestion avoidance. trunking will save some bytes in flight iff one has four or more streams moving between two

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

the freebsd port tree version is dead because of the openh323 issues. before i start hacking, i am hoping someone else has a freebsd version that will build on -current. and i do not care about h232. dare i hope? randy

sip debug shows that my * is trying to invite my spa and being told 404 Reliably Transmitting: OPTIONS sip:42.7.11.194 SIP/2.0 Via: SIP/2.0/UDP 128.9.0.39:5060;branch=z9hG4bK43efe1d7 From: "asterisk" <sip:asterisk@128.9.0.39>;tag=as39d40d19 To: <sip:42.7.11.194> Contact: <sip:asterisk@128.9.0.39> Call-ID:

i am invoking op from a python proggy which does an op.system() of op chmod 640 /usr/local/etc/tac_plus.conf i get "Permission denied by op" % ls -l /usr/local/etc/op.access -r-------- 1 root wheel 149 Jan 13 07:41 /usr/local/etc/op.access % cat /usr/local/etc/op.access # 2007.01.13 # #DEFAULT users=src # chown /usr/sbin/chown $* ; users=src chmod /bin/chmod $* ; users=src

so i just found that one of my hosts is GENERATING these probe pairs, maybe every minute or two (note the sequence numbers): seq my host victim(s) --- ---------------- --------------- 24) 192.168.0.2:1121 <--> 216.52.3.2:2703 25) 192.168.0.2:1122 <--> 216.52.3.4:2703 39) 192.168.0.2:1124 <-->

if the dns has _sip._tcp.my.dom. SRV 0 0 5060 asterisk.dom.ain. _sip._udp.my.dom. SRV 0 0 5060 asterisk.dom.ain. where asterisk.dom.ain. is an A RR of the asterisk pbx. how does a call to sip://user@my.dom come in to asterisk so i can route it? do i just put in sip.conf [username] context = from-url-username and extensions.com [from-url-username] exten =>

>> passdb { >> driver = pam >> } >> passdb { >> driver = pam >> name = pam >> } > Are those two passdb blocks intentional? > > One of them is missing the name parameter. doh. first removed. Aug 3 06:49:23 psg auth: in openpam_parse_chain(): /etc/pam.d/dovecot(1): missing or invalid facility Aug 3 06:49:23 psg dovecot:

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subscribers to the digest form of this list do so in order to only receive the email infrequently. in my case, and i suspect others, twice or so a day would be preferred. the list currently batches about every hour. it is sufficiently annoying that one tends to delete batches. i have written to the list admin about this and received no response, undoubtly they are busy reading the mail :-).

Hi I am using Samba 2.0.6 on a SUN SPARC Solaris 2.6 machine. I compiled Samba using SUN cc with CFLAGS of "-fast". All the remaining configure parameters were left unspecified. I have been trying to use Samba to backup a nerwork of PCs onto the UNIX box. Each PC has a shared network drive, and the user issues an smbtar or smbclient backup command. eg. % smbclient //dcmbpc/C

so, is it safe to put canreinvite=yes on a 7960? on a 1750? on a spa-x000? an xten? how the heck do i find out other than the hard way? randy -- ps: pun intended

instead of exten => 42/1,1,GoTo(hell,s,1) exten => 42/42,1,GoTo(hell,s,1) exten => 42/666,1,GoTo(hell,s,1) exten => 42/69,1,GoTo(hell,s,1) i would like to do something like exten => 42/(1|42|666|69),1,GoTo(hell,s,1) or better use a variable [global] poofs => (1|42|666|69) [foo] exten => 42/${poofs},1,GoTo(hell,s,1) i keep circling the wiki

if a phone number starts with one of 50+ prefixes, i want to send the sip call to gateway X. if it is in any other prefix, i want to send it to gate Y. i am not excited about a looooong list of extens, but will do it if i have to. i suspect there is a database hack, but i lose all database contents if i reinstall the port (this may be a feature of the freebsd port), and i have not figured out a

when i exit more, vi, ... i like to have the window restored to the way it was before i entered more, vi, ... i used to effect this by a hack to the termcap data, but have lost the hack. what am i missing here? randy