similar to: chan_sip + MD5 encryption: WARNING Format for authentication entry is user[:secret]@realm

Anyone had Realtime working with IAX2 and RSA authentication to connect two PBXs, please? It seems that inkeys/outkey fields are not read at all and the following warning is logged when dialing: Jun 2 02:41:36 WARNING[6299] chan_iax2.c: I don't know how to authenticate ******** to XXX.XXX.XXX.XXX Using iax.conf it perfectly works. Maybe a bug in Realtime? TIA, Alex

Finally I decided to rewrite my dialplans according to the right sintax, that is exten => someexten,priority,application(arg1,arg2,...) should be exten => someexten,priority,application,arg1|arg2... Isn't there anybody skilled enough in regular expressions that could write a quick Search 'n' Replace vi command, please? TIA, Alex

Hello, I try to compile the driver zaptel and they give the following error: linux01:/usr/src/zaptel# make install gcc -Iir/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB -Iir/drivers/ l -I. -Wstrict-prototypes -fomit-frame-pointer -Iir/drivers/net/wan -Iir /net -DSTANDALONE_ZAPATA -o zaptel.o -c zaptel.c In file included from zaptel.c:44: /usr/include/linux/module.h:21:

Hi, to register my Asterisk with a SIP provider I use the following syntax, as shown in the default sip.conf: register => 2345:password@sip_proxy where [sip_proxy] type=peer context=from-messagenet host=sip.messagenet.it port=5061 <------------- please note this one!!! 5061 is provider's port I have to register to. This also would work for me: register =>

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

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

ok so my table looks like this... REATE TABLE `sip` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(7) default NULL, `callgroup` varchar(10) default NULL, `callerid` varchar(80) default NULL, `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip`

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

maybe a dumb question but what do we have here??? sip-friends.sql # # Table structure for table `sipfriends` # CREATE TABLE `sipfriends` ( `name` varchar(40) NOT NULL default '', `secret` varchar(40) NOT NULL default '', `context` varchar(40) NOT NULL default '', `username` varchar(40) default '', `ipaddr` varchar(20) NOT NULL default '',

Hello, Right now there is not a table build script at: http://www.voip-info.org/wiki-Asterisk+RealTime+IAX Therefore I have taken the SIP build script and added a few fields that I use from my iax.conf (could be more out there, please see the complete build script below): `dbsecret` varchar(100) default '', `notransfer` varchar(100) default '', `inkeys` varchar(100)

Hello asterisk users, here is a little problem pulling out iax peers from real time database I have the following peer configured in my database mysql> select name,username,secret,type,context,host,disallow,allow,defaultip,deny,permit, ipaddr,port from iax_users where name='iaxtermination'; +----------------+----------+----------------------------------+------+-----

Not being an asterisk expert, but having been around the block once or twice when it comes to data and the like, I have made some observations based on the examples given on voip-info.org Sip configs. it appears there is an adjustment to be made in the sip_buddies example table: >>> name Although set to 30 characters, I don't see where it is limited in the text file. In theory,

Hi, I'm trying to set up an Asterisk pbx based on a Fedora Core 1 Linux box (customized kernel version 2.4.24). I want calls from my SIP soft-phones to simply be dumped onto the PSTN line via a BRI (EuroISDN). I have a cheap HFC-S based PCI ISDN card connected to the NT1+ interface, so I need zaphfc. I've read everything I've found at www.voip-info.org, then I've downloaded the

Hi, Red Hat 9.0 Asterisk 1.2.7.1 Whenever I start Asterisk, I am unable to call out on my SIP channel: >-- Executing Dial("Zap/1-1", "SIP/6137451576@6477235412||t|") in new stack >Apr 30 11:02:00 WARNING[12814]: chan_sip.c:1973 create_addr: No such host: 6477235412 >Apr 30 11:02:01 NOTICE[12814]: app_dial.c:1029 dial_exec_full: Unable to create >channel of type

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

Kamailio has both a ha1 and ha1b column in it's user schema: ha1 = H(A1) = MD5(user:realm:password) ha1b = H(A1b) = MD5(user at realm:realm:password) This is intended to support some devices that append @realm to the user and/or to allow users to put either "user-part only" or "user at domain" into the auth-user field of their UA. Can anybody comment on the following:

I am going to change my Digest realm to match my DNS SVR record. I dug through the code in chan_sip.c and on line 2748 I found it hard coded <frown> : snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", r\anddata); I'm going to change this to : snprintf(tmp, sizeof(tmp), "Digest realm=\"isdn.net\",

Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password.

Dear All, I'm syncronizing a Win$$2k smb share to linux box (centos4). Rsync ver is 2.6.3 in my i18n file you can find: SYSFONT="latarcyrheb-sun16" LANG="it_IT" SUPPORTED="it_IT@euro:it_IT:it:en_US:en" when I try to make the syncronization I've go the followin error on file with special char Eg.: file has vanished:

Hello, they are successful to start asterisk, task that the error that I had previously had had to a configuration problem. Start asterisk in modality consol and when two softphone speaks is not felt well, and I have the following error: -- Registered SIP '1000' at 10.0.0.7 port 5060 expires 1800 -- Saved useragent "X-Lite release 1103m" for peer 1000 --