similar to: 2 line SIP ATAs with Asterisk using RealTime

Hi all I'm trying to figure out what values are valid for the "insecure" option in a realtime configuration table. The table field is 4 chars long and the actual valid values for this is longer. Can I modify the field length or has this changed? Below is where I looked, if I'm not looking in the right place please let me know. the field on the table is: ... `insecure`

Hello We have recently upgraded to Realtime engine (sip buddies and extensions) and now have problems with calling local SIP users. I have rtcachefriends=yes but tried with 'no' and it's even worse. (asterisk 1.8.1.1 + realtime mysql) Here's an example: User 1000 registers successfully and can then be called with Dial(SIP/1000,30) successfully After some time when I try to call

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]

Asterisk: 1.4.17 with sip realtime Openser 1.3.x Hi, I had this setup working fine until I try putting OpenSER in the picture as a proxy. Unauthenticated calls go to a PRI based app via a ZAP channel, calls to sip users get send to them etc. Now with a proxy in the picture asterisk asks the incoming calls for authentication "407 Proxy Authentication Required". It seems that the

Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]

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 :

Hi, I have installed asterisk real time and sip buddies information is being stored in DB. Now I have a question, Asterisk Realtime Server -A Third party SIP server-B Question: Is there any configuration in * RT that it can register with defined sip user on Server-B I was only able to find sip users information in DB not about user registration on other server. -ag -------------- next part

Hi! I use the following configuration to register my asterisk server to my SIP provider: register => 12345:passwd@sip.provider.com/12345 sip.conf: [sipout-test] type=peer username=12345 fromuser=12345 fromdomain=provider.com secret=passwd insecure=very host=sip.provider.com qualify=yes context=test-incoming extensions.conf: exten => 12345,1,Dial(SIP/10) exten =>

I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a

Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

I'm new to asterisk and need some help with getting a SIP connection working. I am trying to establish a termination point/DID number in another country. I am currently running Asterisk CVS-HEAD. My foreign provider uses SIP and authenticates via IP address. I am not required to register my SIP connection in order to send or receive calls. Can someone help me with how to understand the

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov

Hi All, I am trying to configure asterisk realtime. But i am unable to get the extensions listed successfully when i type "sip show peers" in the asterisk CLI . i am unable to see any failure logs when i do a reload i can able to connect to the data source through "odbc show" in the CLI, Any hep in this regard is highly appreciated. Following is the configuration

Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly it does not register but I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 message

Hello all! My basic problem is that we haven't been able to get realtime to use ODBC to store configuration data. Here are the details: We've installed Asterisk on a CentOS machine as follows: 1. Downloaded, compiled, and installed FreeTDS 0.63 2. Downloaded, compiled, and installed unixODBC 2.2.11 3. Downloaded, compiled, and installed Asterisk, Asterisk-Addons, and Zaptel from CVS

Hi all, I registered a SIP account at budgetphone.nl/talkin2ya.nl Receiving calls works like a charm, I even redirected my normal PSTN number to the number I got from them so everything ends up in my * server. Before I ask them to take over my normal phone number I wanted to test all of it, so I ordered some calling minutes to test. Now I cannot get outbound calling to work with them. Anyone here

Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

First of all, if Asterisk is the client and it must register to the other side, does the peer\user entry have to be in sip.conf, or can it be in ARA? Second, why do all calls fall through to the last context specified, whether in that peer\user definition or not? I'm assuming it's a typo somewhere, but I can't find it. I had a full sip.conf, but axed a lot of the fluff trying to