search for: disallow

In my sip.conf, under general I have: disallow=all allow=g729 allow=alaw allow=ulaw Then I have a specific sip: [RNK] <clip> disallow=all allow=alaw allow=ulaw allow=gsm If I do this: exten => _9.,1,Dial(${EXTEN}@RNK,60) The call still goes out as G729 even though I've told the RNK to disallow g729. I need to be able to make o...

...ind any solution. First I thought this was a problem with the phones, but all my phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried all firmware versions I could find for the phones. First, my situation: - No NAT, No Firewall, same subnet - Codec configuration: In general: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw In the phones: disallow=all disallow=g723.1 disallow=g729 disallow=gsm allow=ulaw allow=alaw But I also tried other codec configs. (allow=gsm, etc). Same problem. I'm testing from the Cisco 7960, as this phone seems to work...

...he final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and ulaw) - PSTN Peer (supports g729 and ulaw) - Remote Asterisk Peer (supports speex and ulaw) Currently, it's configured like this: [snom300] disallow=all allow=ulaw [pstnpeer] disallow=all allow=ulaw [asteriskpeer] disallow=all allow=speex which translates to this: Snom300 ---ulaw---> (pass-thru) ---ulaw----> PSTNPeer Snom300 ---ulaw---> (transcode) ---speex---> AsteriskPeer In other words, my Snom phone al...

We want to record inbound channels at 16kHz, but send only 8kHz to our peers. I've set our default profile in sip.conf to disallow all but g722, and the peers disallow all but ulaw. We have a proxy in front of Asterisk that is configured to disallow all but G722 also. My test calls show inbound to the proxy is recorded at 16kHz, inbound in Asterisk is only 8kHz, and the peers receive 8kHz. So the only thing not working is Ast...

Is there a straightforward way to disallow the deletion of all IMAP mailboxes? I have a user who's deleted an important IMAP mailbox and I'm now recovering a recent copy from the backup. But I'd rather just blanket disallow all folder deletions. The user is using Thunderbird and this has happened more than once so I suspec...

On Wed, Oct 13, 2010 at 11:10 PM, Anton Korobeynikov < anton at korobeynikov.info> wrote: > > indexing the llvm.org svn archive. This means that when you search for > an > > LLVM-related symbol in code search, you get one of the many (possibly > > out-of-date) mirrors, rather than the up-to-date llvm.org version. This > is > > sad. > This is intentional. The

...=> <my UserID>:<my password>@<ITSP A Server #1 domain> register => <my UserID>:<my password>@<ITSP A Server #2 domain> register => <my UserID>:<my password>@<ITSP B #1 domain> notransfer=yes bindport=4569 bindaddr=0.0.0.0 bandwidth=low disallow=all allow=ulaw allow=g729 jitterbuffer=yes forcejitterbuffer=no tos=lowdelay autokill=yes [ITSP_B] context=incoming-iax type=friend qualify=2000 host=<ITSP B #1 domain> user=<my UserID> username=<my UserID> auth=md5 secret=<my password> disallow=all allow=ulaw ; ; *** ITSP_...

Hi, I'm receiving the following messages in my mail logs that I haven't seen before: Nov 28 22:45:31 bwipropemail dovecot: auth: login(?,179.210.41.21): Username character disallowed by auth_username_chars: 0x13 (username: AB?) Nov 28 22:45:31 bwipropemail dovecot: auth: login(?,179.210.41.21): Username character disallowed by auth_username_chars: 0x13 (username: AB?) There's thousands of them, from hundreds of different IP addresses. I suspect it's an exploit attemp...

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155 Anybody

...DHCP gets an address, loads the SIP software and sets there for me to dial. However, I get the INV when I dial. Any ideas on why the phone is displaying invalid and what to do about it??? Thanks, jerry sip.conf ------------------------ [201] type=friend dtmfmode=rfc2833 username=201 secret=201 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media Assistant" <201> [202] type=friend dtmfmode=rfc2833 username=202 secret=202 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media Assistant" <201> [203] type=frie...

...can register with each other when both firewalls are blocking port 4569? Do I have to open ports 4569 on both servers in order to register them with each other? ======================== SERVER 1: register => Ast-1-in:password1@IP-address [Ast-2-in] type=user secret=password2 context=incoming disallow=all allow=ulaw ;trunk=yes [Ast-2-out] type=peer disallow=all allow=ulaw host=dynamic ========================== SERVER 2: register => Ast-2-in:password2@IP-address [Ast-1-in] type=user secret=password1 context=incoming disallow=all allow=ulaw ;trunk=yes [Ast-1-out] type=peer disallow=all all...

...one. Of course I considered setting the allow statements in the sections of sip.conf. But it seems these settings are only considered on incoming calls, but not on outgoing calls (initiated by a Dial application with the sip.conf section as its argument). So, when I have something like [general] disallow=all allow=alaw allow=ulaw allow=ilbc [SipgateAccount] type=friend ... disallow=all allow=ilbc then these settings are only considered when I get an incoming call, but not when I make an outgoing call using "exten=>...,Dial/SIP/${EXTEN}@SipgateAccount,60,)" Thanks in advance, Andy -...

...Mask Port Status ast33 192.168.42.130 (S) 255.255.255.255 4569 (T) OK (30 ms) Here's my info: ast551: 192.168.1.130 ast33: 192.168.42.130 Version: CVS-HEAD-11/03/04-14:59:37 (both boxes) IAX.CONF on ast551: [general] bindport=4569 notransfer=yes disallow=all allow=ulaw [ast33] type=friend auth=md5 secret=pass context=no-callwaiting host=192.168.42.130 qualify=yes trunk=yes disallow=all allow=ulaw IAX.CONF on ast33: [general] bindport=4569 disallow=all allow=ulaw [ast551] type=friend auth=md5 secret=pass context=no-callwaiting host=192.168.1.130...

Hi, On the general section of my sip.conf I had a disallow=all. Then I put disallow=all, allow=g729, allow=ulaw on my users. It didn't work until I removed the disallow=all from the header. I know disallow=all in the header is totally useless in this case (since I have it for every user), but anyway, is this the correct behavior? Thank you Dov ----...

...I get this in CLI: -- Reloading module 'app_playback.so' (Sound File Playback Application) [May 3 20:04:26] NOTICE[13892]: app_playback.c:455 reload: Reloading say.conf == Parsing '/etc/asterisk/say.conf': Found [May 3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' [May 3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' [May 3 20:04:26] WARNING[13879]: frame.c:1289 ast_parse_allow_disallow: Cannot disallow unknown format '' [May 3 20:04:27] WARNING[...

Currently I'm using a linux+samba+PPP I didn't had to do something special to disallow samba over the PPP connection. Must I do something to disallow samba over a second NIC (=not the LAN) or how must samba be bound to the nic of the LAN ?

Hello Everyone, I have disallow=all and allow=g729 set in sip.conf however, it seems that asterisk still thinks it support other codecs: Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw). How can I disable gsm,ulaw,alaw..... Thanks in Advance, Nick.

> On 11/02/2017, at 3:40 am, Frank Vanoni <mailinglist at linuxista.com> wrote: > > On Thu, 2017-02-09 at 14:58 +0200, ????? ?????? wrote: > > >> so the main question is -- how to Disallow CALLS without registering >> on PBX > > sip.conf configuration > In the [general] section, define: > > > [general] > ... > allowguest=no > alwaysauthreject=yes > ... I don't think either of these configuration attributes achieve what the OP requested (&qu...

...nf to limit ; to one inbound or outbound call per phone. Use ; the group counters in the dial plan for that. ; ;mailbox=1234 at default ; mailbox 1234 in voicemail context "default" disallow=all ; need to disallow=all before we can use allow= allow=ulaw ; Note: In user sections the order of codecs ; listed with allow= does NOT matter! allow=alaw allow=g723.1 ; Asterisk only supports g723.1 pass-th...

Hi: It's the first time I have this problem. No matter how I configure my two IAX machines the connection is rejected. "chan_iax2.c:5550 socket_read: Call rejected by ****: No authority found" iax server A: [saad_out] type=peer host=hostip username=username secret=secret disallow=all allow=gsm iax server B: [guest] type=user username=username secret=secret context=tele disallow=all allow=gsm Any suggestions of why the connection is refused. I have no firewall. Thanks ____________________________________________________________________________________ We won...