similar to: Making "sip show channels" show sane results with sipfriends from mysql?

Greetings, Is there a way to tie a specific sip username to a IP address when authenticating against mysql sipfriends table? (USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 in channels/Makefile) The reason is that I'm using Wellgate FXSes that have second/third/fourth FXS ports bugged when I use a password, but work ok when there is no password. Linking the username to a specific ip could

Hi all, I am new to asterisk and I can not find any detailed info on using SIP MySQL support (sipfriends) with clients behind NAT. I've heard that I have to patch chan_sip.c and Makefile to get it working. I tried on voip-info.org but found no answer for my questions. I found some answer on Digium mail list archive: http://lists.digium.com/pipermail/asterisk-cvs/2004-January/000854.html

hi using the following in sip.conf, codec preferences aren't set, and asterisk uses alaw whatever I do, except force it to one specific in the [user] [general] disallow=all allow=g726 allow=g729 allow=gsm allow=alaw then, from 'sip show peer something' it tells me Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (none) can someone please explaing why? this is

Hi, I have some mySQL-sipfriends and connectivity to PSTN. When a call from PSTN comes, it shows a callerid, and that callerid is displayed at the called sip phone. When the call comes from another sip user (defined as mySQL-sipfriend), no callerid is displayed at the called sip phone. I turned on sip debug and discovered, that in the last case in the SIP-header to the called phone: From:

After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table. == Binding sipusers to mysql/asterisk/sip == Binding sippeers to mysql/asterisk/sip Feb 17 15:20:03 WARNING[15317]: config.c:823 read_config_maps: The

Hello I was wandering If I let 2 asterisk boxes (let's name them ast01 and ast02) connect to one SQL realtime iaxfriends/sipfriends database What happens if I register my client to ast01, The ast01 box will update the client's record in the iaxfriends database (ipaddr/port/regseconds) Let's say there is an incoming call then for this client but this call arrives on ast02 (the box

I'm authenticating against sipfriends in MySQL, and have just noticed that none of the below commands return any output: sip show users sip show inuse sip show active sip show subscriptions Is this a bug or something wrong on my side? I'm using the stable 1.0 cvs Vahan

hi Username : 1000012 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (gsm|g729|g726|alaw|ulaw) the above is from SIP SHOW PEER 1000012, and as it clearly shows, g.729 is preferred before alaw. If I dial this SIP - * - SIP from a phone with G.729 enabled, it uses G.729. However, if I dial from my cell phone - GSM - PSTN - * - SIP, the call uses ALAW, which I thought it

Ok, With everything restore on rtp.c, still I have no audio however the call is not destroyed immediately as before. I'm going to put a second Granstream box, and findout if between two boxes this happen too. I cannot believe that we cannot do 2 g726 on the same box at one time. Carlos -----Original Message----- From: Carlos Alperin [mailto:calperin@senecacom.net] Sent: Wednesday,

Hi, I am running FC1 with latest patches of 3/30/04, and I have the latest CVS as of this morning 3/30/04 of asterisk, zap and libpri. The SIP device I am using is a Sipura SPA-2000 with G726-32 "Forced". When I 'make clean" and recompiled zaptel, libpri, asterisk and start asterisk I can see: [format_g726.so] [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) ==

hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version. i like to store usernames and passwords in a sql database. i like to log failed authentification-passwords, to create a blacklist for securityreasons. i thingk a sql-database is a good way to log these actions. i don.t find debugging-options to output invalid login-passwords. Ok, i have made the following: debian is my OS.

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi, I have both codec_g726.so and format_g726.so loaded: root at test:~# asterisk -r -x "module show" | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE ..... [SIP headers omitted] v=0

Dear all, I'm trying to use ulaw and alaw with Diax and Asterisk but I'm not able to, I got the following error message: Mar 20 11:47:59 NOTICE[7099]: chan_iax2.c:6350 socket_read: Rejected connect attempt from 192.168.0.55, requested/capability 0x8/0xc incompatible with our capability 0xfe02. I do not understand why because my Asterisk box load these codecs properly! Does somebody

Hello there, I have configured my asterisk to run on Mysql backend. But the Asterisk was unable to pick the peer details from the database. This is how I configured the Asterisk to run with mysql on the backend. Edit /usr/src/asterisk/channels/Makefile, change it to enable the MYSQL_FRIENDS USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 cd /usr/src/asterisk make

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

Dear all, I am doing an interop testing with asterisk-1.6.0.5 now, and I have a question about the G.726 codec on asterisk. While my IAD supportes G.726-16,24,32 and 40 codecs, when doing a testing about G.726-40, I found that asterisk removed the G.72-40 sdp attrib when transmitting the INVITE with SDP. I modified sip.conf in order to solve the problem, G.726-32 is ok when allow=g726, but

hi all the help output for DISA ends like below, with the half-sentence 'Note that in the case' what's the rest of that sentence? The file that contains the passcodes (if used) allows specification of either just a passcode (defaulting to the "disa" context, or passcode|context on each line of the file. The file may contain blank lines, or comments starting with

Since nobody answer my previous question (It looks like g.726 is a bad word). I have this scenario: One box with Asterisk 1.4.0 beta 2 IAX to anothers Asterisk working properly. As an ATA I have only one Grandstream HT496. Two lines on the ATA 727 & 726. >From outside I can call any of those two extensions if: I defined both as ulaw (G.711) One as ulaw and the other as G.729

Hello All, I was wondering if somebody could elaborate the change in translation of codecs specifically the amount of time increased in Asterisk 11. For example *Asterisk 11* * **alaw **speex * *gsm **15000 **15000 * *ulaw 9150 15000* * * *Asterisk 1.6.x* * **alaw **speex * *gsm **2 12002 * *ulaw 1 12002* I did recalculate the