similar to: How to append custom option to Contact: header on outgoing SIP INVITE msgs?

Hi, A brief 6-step guide on how to hardcode a change in the Asterisk source that will allow it to work from behind a nat device. I know it?s messy, but it may prove useful to some people. 1. First punch a whole in your nat device. I just forwarded the port 5060 (for sip) and all ports between 10000 to 10020 (for rtp) to my asterisk gateway. 2. Now make sure your /etc/asterisk/rtp.conf correctly

Hello, If I understand it correctly, when sending INVITE, o= and c= sdp fields are built using p->ourip IP address. At this point RTP packets will be coming to the default asterisk IP address. For the machine with multiple interfaces this could be not the right one (not what we want). Could it be configured (in rtp.conf or in sip.conf per context) ? Thank you. Alex Zarubin --------------

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your

Hi I have a problem with Asterisk-1.6.0.3-rc1 and manager API. I want to dial out from manager's console and with Asterisk 1.4.X this settings were OK. Action: Originate Channel: SIP/384 Context: main Exten: 102 Priority: 1 Callerid: 384 I could dial out, but with asterisk 1.6 I get this error. Response: Error Message: Channel not specified I have originate and system privilege in

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux

I've been banging my head on a brick wall for about an hour now trying to understand why the following doesn't work (which is even provided as an example in the distribution!). The goal is to create a voicemail-only extension not associated with a phone. I'd rather not have an extension dedicated to VoicemailMain(), so I would like the user to be able to hit '*' during

Hi, I need help with modifying cdr_addon_mysql.c I want to have more fields in cdr table in asterisk. I've tried to modify cdr_addon_mysql.c and replace userfield with ex team (sed -e 's/userfield/team/g' ). When I try to recomplie menuselect/menuselect --check-deps menuselect.makeopts Generating embedded module rules ... make[1]: Nothing to be done for `all'.

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

I'm getting this message when I start Asterisk chan_capi.c:2205 capi_handle_msg: Command.Subcommand = 0x5.0x81 but when I try and recompile I get this chan_capi.c:60: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) any help would be greatly appreciated. -- Dave Cotton <dcotton@linuxautrement.com>

there is a problem in compiling asterisk-addons any one have fixed this problem. i want res_config_mysql.so any one help me ----------------------------------------------------- [root@localhost asterisk-addons]# make cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o res_config_mysql.o res_config_mysql.c res_config_mysql.c: In function `realtime_mysql':

Hi ALL; I got the latest Asterisk-addons for Mysql-Cdr, but I have problem compiling that.It says: # make ..... ........ res_config_mysql.c: In function `realtime_mysql': res_config_mysql.c:143: warning: passing arg 1 of `ast_strlen_zero' makes pointer from integer without a cast res_config_mysql.c: In function `realtime_multi_mysql': res_config_mysql.c:242: warning: passing arg

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad this patch

hey guys I thought I was making progress on my dialplan when I realized that the class features that are available for zap channels aren't available for SIP channels. I see references in the archives to adding pattern matches in the dialplan for CLASS features which has raised a couple questions. 1. Is implementing CLASS like features via the dialplan the currently recommended way to do

Revision 77616 Modified Sat Jul 28 07:44:16 2007 UTC (3 months ago) by rizzo File length: 681368 byte(s) Diff to previous 77538 make use of received= and rport= fields in sip replies. In a nutshell, these fields are used to tell a sip entity the address and port its request came from, and are extremely useful in the presence of NATs, especially with symmetric NATs where STUN is totally

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number