similar to: What are the minimum realtime fields for sipusers?

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is "403 Forbidden". Some people told that asterisk is not sending the username in the OPTION, required by the pstn. Taking a look of the example of rfc3261.txt

Hello using asterisk 1.8.32.3 I am not able to make a call with video support. I do not know what I am missing to make this video call. Codec h264 should be supported. sip*CLI> core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INT BINARY HEX TYPE NAME

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

Hi, I have 2 databases DBNAME1 and DBNAME2. In each database I have a table called sipusers (so DBNAME1.sipusers and DBNAME2.sipusers). Can I use both sipusers tables in Asterisk RealTime? Something like this: /etc/asterisk/extconfig.conf: [settings] sipusers => odbc,DBNAME1,sipusers sippeers => odbc,DBNAME1,sipusers sipusers => odbc,DBNAME2,sipusers sippeers =>

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hi. we are experimenting a strange issue in our PBX. By example: if we dial to the 100, the call is answered in 199. We dont have any redirection for that, but the cli show the same issue when request show peers. Aditionally, the user 100 use the ip address 192.168.11.100, and the cli show connected the user from 192.168.11.160 (that ip is assigned to the user 199) PBX*CLI> sip show peers

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

Hi After recently upgrading to 1.8.3 I have noticed that the nat setting for my peer in my sip table is not making it into the realtime cache. For example * Name : 501 Realtime peer: Yes, cached Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pack-local Subscr.Cont. : <Not set> Language : AMA flags :

I wasn't able to get much out of babytel, beyond the fact that I was, apparently, sending options which is why I'm not getting 200 OK. How can I, generally speaking, ping/telnet or otherwise test the connection to get more data? A connection to this peer directly from a softphone, Jitsi, works fine. linux-k7qk*CLI> linux-k7qk*CLI> sip show peer testcarrier * Name :

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than "NOT_INUSE". I have two extensions: 6666 and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use 6666 to call 6668 and in the dialplan have a noop to see what

I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a starfish on it. In some ways, astonishing that it's not really that definitive, it's more general -- and it only clocks in at one ream of paper! In any event, I'm having some port problems on my home network: http://security.stackexchange.com/questions/81752/ I need to open ports for

Hi all, From the web, I can find a table scheme of sipusers for ARA using. However, I can't find any meaning of each field, especially for the field regserver which is new in the table. Can any tell me more detail about the usage of each field? CREATE TABLE `sip_buddies` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `host` varchar(31) NOT NULL

Hi, I would have expected that peers of type friend ( for example an SIP-phone) registring at Asterisk will be searched in sipusers. But the peers will be searched in sippeers. May be sombody can explain the difference? Asterisk 1.4 thanks Thomas

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on

I have followed the instructions on voip-info.org for Realtime SIP peers, but I get this notice : [Feb 22 20:05:32] NOTICE[15298]: chan_sip.c:15889 handle_request_register: Registration from '<sip:testsip at 192.168.1.150;transport=UDP>' failed for '192.168.1.105' - No matching peer found The CLI shows : [Feb 22 19:58:23] == Parsing

On 2/20/15 6:15 AM, thufir wrote: > What's the difference between user "123" and "devries"? Based on the > output here, they seem the same..? > > tleilax*CLI> > tleilax*CLI> sip show users > Username Secret Accountcode > Def.Context ACL Forcerport > 201 password 201 > default

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

hi folks, I am using asterisk 1.2.13 (debian etch). My customer's sip accounts are stored in realtime sipusers. I have enabled in sip.conf rtcachefriends=yes and ignoreregexpire=yes Each account has nat=yes Now, I have lot of problems. for example, when I change the 'secret' field of a user in the database, it doesn't get reflected in Asterisk, who is still expecting the old