Hi,
I've got a problem with openser servers and load balancing. Current setup contains 2 asterisks and 2 opensers. I've got a srv record xxx for both asterisk-1 and asterisk-2 - same priorities and weights.
Openser's config runs basically:
rewritehostport("xxx");
t_relay();
But all packets go to first host on the list anyways. For example now I get this from dns (same order):
_sip._udp.xxx. 300 IN SRV 1 50 5060 asterisk-2
_sip._udp.xxx. 300 IN SRV 1 50 5060 asterisk-1
and got 6 calls on asterisk-2 and none on asterisk-1. If I change the order, all calls go to the first host on the list. Shouldn't openser do load-balancing in this scenario? I don't use any options in t_relay().
Thanks for ideas.
Hi,
I am facing mediaproxy problem during Nated-pstn calls.
Here is the problems:
1. Mediaproxy session has been broken
2. No called address on mediaproxy server
3. Mediaproxy session status is inactive
4. No bytes has been passed through mediaproxy
So please help me to solve all this problems.
Thanking you in advance..
- Krunal Patel
I figured out the cause of the errror.....
Its funny.....on linux you execute a command and parameters like this
sayIt helloworld right..
but on suse you execute a command like this
./sayit helloworld
The only exception is if you specify a full path to sayIt like this
/usr/bin/sayIt helloworld
so in the ser_mysql.sh file gen_ha1 should be changed to ./gen_sha1 and
bingo.....no errors....
This same thing goes for all other commands that generate command not
found error...it actually happened with serctl too and i fixed it
Thanks for listening
Akintayo Olusegun
Access Solutions
Greetings and regards to Everyone.
I'm running Openser/Mediaproxy with NAT traversal. Opeser initially was
running on port 5060. Our client is behind Cisco firewall and at some point
they turned on Sip fixup (which is beyond our control ) because they need
sip fixup for some other application and as a result nat travesal can't
work. Per my conversation with their IT guy they have no problem with our
sip proxy running on port 5070. I've reconfigured proxy on port 5070 -
everything is working with no problems. Here are my questions:
1. Anything wrong running sip proxy on port 5070 or some other port (if it
does not interfere with other services) in general.
2. Would such solution considered as permanent solution?
TIA
Toly
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Question . .
I'm trying to use the load_gws with a particular group id per the docs for 1.3, and am getting the following error message when I try and start OpenSER:
May 1 20:40:05 [16853] DBG:core:fix_actions: fixing load_gws, line 487
May 1 20:40:05 [16853] ERROR:lcr:fixstringloadgws: Wrong value <2> for param <1>!
May 1 20:40:05 [16853] ERROR:core:fix_actions: fixing failed (code=-1) at cfg line 487
May 1 20:40:05 [16853] CRITICAL:core:fix_expr: fix_actions error
May 1 20:40:05 [16853] ERROR:core:main: failed to fix configuration with err code -1
May 1 20:40:05 [16853] INFO:snmpstats:mod_destroy: The SNMPStats module got the kill signal
May 1 20:40:05 [16853] INFO:snmpstats:mod_destroy: Shutting down the AgentX Sub-Agent!
May 1 20:40:05 [16853] DBG:xlog:destroy: destroy module...
May 1 20:40:05 [16853] DBG:tm:tm_shutdown: tm_shutdown : start
May 1 20:40:05 [16853] DBG:tm:unlink_timer_lists: emptying DELETE list
May 1 20:40:05 [16853] DBG:tm:tm_shutdown: emptying hash table
May 1 20:40:05 [16853] DBG:tm:tm_shutdown: releasing timers
May 1 20:40:05 [16853] DBG:tm:tm_shutdown: removing semaphores
May 1 20:40:05 [16853] DBG:tm:tm_shutdown: destroying callback lists
May 1 20:40:05 [16853] DBG:tm:tm_shutdown: tm_shutdown : done
May 1 20:40:05 [16853] DBG:core:shm_mem_destroy:
May 1 20:40:05 [16853] DBG:core:shm_mem_destroy: destroying the shared memory lock
Am I missing something? Per the docs at http://www.openser.org/docs/modules/1.3.x/lcr.html#AEN303 , I should be able to use a particular gateway group id in that function.
My config code excerpt:
if (from_gw("1000")) {
xlog("L_ERR", "Route 14: Call originating from incoming vendor\n");
if (!load_gws("2")) {
xlog("L_ERR", "Route 14: Error loading gateways (GW Group 2) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("503", "Termination Currently Unavailable");
exit;
}
if (!next_gw()) {
xlog("L_ERR", "Route 14: No gateways available (GW Group 2) - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
sl_send_reply("503", "Termination Currently Unavailable");
exit;
}
setflag(21);
t_on_failure("1");
route(10);
}
Basically, it's checking to see if the call is originating from one of the incoming gateways in group id 1000, and if it is, I'd like to pull a gateway from group 2 and send the call to it.
Any help would be appreciated.
Thanks,
- Darren
Hi,
I am facing mediaproxy problem during Nated-pstn calls.
Here is the problems:
1. Mediaproxy session has been broken
2. No called address on mediaproxy server
3. Mediaproxy session status is inactive
4. No bytes has been passed through mediaproxy
So please help me to solve all this problems.
Thanking you in advance..
- Krunal Patel
Hi guys,
I'm trying to set up a SER server between 2 asterisk machines. I run
into 2 issues.
Whenever I call someone I don't get any ringback tone even so the call
initiating asterisk machine gets the 180 message after 100.
<--- SIP read from 10.4.1.80:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060
From: "Thorsten" <sip:1000@82.98.89.134>;tag=as4c964973
To: <sip:017683035400@10.4.1.80>
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)82.98.89.134
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.7 (i386/linux))
Content-Length: 0
Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459
req_src_ip=82.98.89.134 req_src_port=5060
in_uri=sip:017683035400@10.4.1.80
out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1"
<------------->
--- (9 headers 0 lines) ---
mg03*CLI>
<--- SIP read from 10.4.1.80:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.4.1.80:5060;branch=z9hG4bK2729bab8;rport=5060
From: "Thorsten" <sip:1000@82.98.89.134>;tag=as4c964973
To: <sip:017683035400@10.4.1.80>;tag=59cea6e4c6ca71e2f82c9c3c8b464af6.bec2
Call-ID: 5e209fbb7ebdbad97f0193515c5a2982(a)82.98.89.134
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.7 (i386/linux))
Content-Length: 0
Warning: 392 10.4.1.80:5060 "Noisy feedback tells: pid=459
req_src_ip=82.98.89.134 req_src_port=5060
in_uri=sip:017683035400@10.4.1.80
out_uri=sip:017683035400@192.168.13.102:5060 via_cnt==1
On SER I've configured to send this message:
if (method=="INVITE") {
if (uri =~ "sip:0[0-9]@*") {
route(3);
sl_send_reply("180", "Ringing");
break;
}
};
The other issue is that I don't see the caller id on the receiver side.
I don't know if it is a asterisk or a SER issue. Only if I set the
caller id on asterisk manual in extensions.conf with
exten => _X.,1,Set(CALLERID(num)=06965006100)
I'll see the caller id on the receiver side.
I would really appreciate any help
Thanks
Thorsten
