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After updating to the latest CVS Feb 17 15:20:03 WARNING[15317]: config.c:819 read_config_maps: The 'sipfriends' table is obsolete, update your config to use sipusers and sippeers, though they can point to the same table.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 all! Has anyone expirienced the following:? With an IAXclient softphone (like diax/iaxcomm/etc) Dialing to the PSTN (zap) or a SIP device has no problems .. but when I make calls between 2 softphones I have weird problems.... in about 4 out of 10 IAX-2-IAX softphone calls I get a big delay .. in the beginning of the call it's all okay... (delay < 0.5 sec) but the longer the call

Hi When I register a SIP or IAX client to asterisk and I dial to it from another UA then there is no problem at all But, when I register two or more clients to the SAME peer (with the same user/pass) and I call to this peer.. Then only the UA which registered the last will ring.. Others don't ring... What can I do about this?? I would like to register for example 10 UA's to the same

Hello maybe someone can help me? I did the CVS checkout and then compiled asterisk Then I tried to compile the addons and got the following (don't understand what's wrong at all and can't find anything about this error on google/wiki) app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but only 3 given app_addon_sql_mysql.c: In function

Hi, The "all" zone you can use in /etc/shorewall/policy isn''t valid in /etc/shorewall/rules, is this correct? I was entering a rule to (for example) block all TCP port 12345 traffic from all sources to all destinations, and logically thinking I began typing this line. REJECT all all tcp 12345 But it didn''t work :-) If I have to enter the zone names, I would

Hello all! Is there any possibility to determine the IP address of a caller in my dialplan? I would like to have a predefined channel variable like ${CALLER_IP} but it seems it doesn't exist (http://www.voip-info.org/wiki-Asterisk+Variables) .. is this list complete? Are there any other possibility to store the SIP/IAX caller's IP address on every call? Thanks Niels

Hi .. When I use the GoTo statement in realtime to goto a priority only ... E.g. Goto(3) then there's no problem But, If I try to jump to another context ... E.g. Goto(othercontext,${EXTEN},3) then it doesn't work If I process the same statement in extensions.conf things go well Are there things broken regarding GoTo in combination with Realtime Extensions ?

How do I allow traceroute to reach my server? Pings work fine but traceroute stops at the last hop before my server. If I shut off the firewall it reaches it fine. PING danicar.net (24.222.246.120): 56 data bytes 64 bytes from 24.222.246.120: icmp_seq=0 ttl=237 time=104.0 ms 64 bytes from 24.222.246.120: icmp_seq=1 ttl=237 time=74.9 ms 64 bytes from 24.222.246.120: icmp_seq=2 ttl=237 time=90.6

I have 2610 XM with 1 Fastethernet and VIC2-2BRI. Dialin and dialout over pots is ok. Also inbound pots calls get redirected to Asterisk y.y.y.y So far so good. But I want to setup VOIP sessions with local carrier. I added dial-peer 40 for this. Session target x.x.x.x But calls will always get routed to the pstn peers 50 and 60. Peer x.x.x.x is never contacted or tried. My situation: PSTN

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

This round implements all of the review comments from V2 and all patches are now acked. Unless there are any objections I intend to apply later this morning. Ian.

First off, Apologies for the massive patch series... This series boots a 32-bit dom0 kernel to a command prompt on an ARMv8 (AArch64) model. The kernel is the same one as I am currently using with the 32 bit hypervisor I haven''t yet tried starting a guest or anything super advanced like that ;-). Also there is not real support for 64-bit domains at all, although in one or two places I

Hi all... I have a slight problem with getting speex running I Downloaded Speex sources (v. 1.0.4 stable version) and did make; make install sucessfully Then I re-maked the asterisk sources and clearly saw a speex.so module being built (so the makefile for sure detects that there is a speex lib installed now) After that when I run asterisk: [codec_speex.so] Feb 22 09:32:59 WARNING[29189]:

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

Hello.. I have configured asterisk to send CDR's to an ODBC datasource on IAX calls I can find the IP address of the caller in the 'channel' field For example: IAX2/<username>@<ipaddr>:4569-458 On SIP calls I never see the IP address of the caller For example: SIP/<username>-9d51 So on SIP calls there is not any possibility to log the ip adress of the

Hello.. How is this possible?? I have 65 active calls .. but making new calls and registering isn't possible anymore the CLI command restart now didn't even work .. had to kill the process before it worked again... myasterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) 65 active call(s) Jun 27 16:22:06