search for: disallowing

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 other 729 calls but to this one paticular group, they

Hi All, I have been busy with this problem for a while now, but I can't find 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

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

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

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 suspect Tbird is willing to let a folder get deleted too easily.

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 out-of-date) mirrors, rather than the up-to-date llvm.org version.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

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

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

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

Hi all, I have setup my Cisco 79XX phone. Did the tftp, put the config files in the right location with the right names. Booted my phone, it does the tftp things, the screen shows my extensions everything seems fine. However, when I come offhook and try to dial 11 which is just a playback of demo-congrats in the dialplan the phone says Calling Out (INV) below is my sip.conf file - I presume it

Here is my configuration everything would seems be straight forward, but I can not register both asterisk with each other. Both asterisks have Static IP but they are behind firewall/router, so it means I have to use Register statement. I'm a bit confused about the register statement. How can they can register with each other when both firewalls are blocking port 4569? Do I have to open

Hi all, Is there a possibility to set the codecs Asterisk will choose in the dialplan ("exten=>" statements or their contexts) instead of sip.conf? My problem is that I connect my SIP phone with several providers (Nikotel, Sipgate, Stanaphone) for icoming and outgoing calls. Not all of these providers offer the same set of codecs. I'd like Asterisk to use the same codec for the

Hi All - Well, after happily existing in a one office environment with asterisk for a few months, I've now decided to start adding in our other offices with their own * boxes and IAX connections (over VPN). Unfortunately, I'm an idiot and I can't get it to work. I'm having some kind of problem with codecs, I guess, but I don't understand what or why. When trying to use

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 -------------- next part -------------- An

Hello, everyone. I've installed asterisk SVN-branch-1.4-r62942 and every time I reload asterisk 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

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 >

Hi all, I need to test the following scenario: +-----------+ +-----------+ | asterisk 1| | asterisk 2| +-----------+ +-----------+ | | | | _______|__________________|___________ | | | | | | +-------+ +-------+ | ATA 1 |

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