[asterisk-users] Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
function count_exec(struct ast_channel *chan, const char *data).But at
compile time char * data cause core dumped. Asterisk-11.5.1 Centos6
app_confbrige.c confbridge.conf
=
Task: Using Dailplan user want to retrive no of user in conference
'6050' = 1. Verbose(3,testMyConfbridgeCount) [pbx_config] 2.
MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,== ${count} )
[pbx_config]
issue:Currently asterisk core dumped as soon as app2 load .
file: app/app_confbrige.c
=
partial code of app_confbridge.c:
static const char *const app2 =MyConfbridgeCount;
static int load_module(void)
{
ast_verb(3 ,==Inside load_module==);
ast_verb(3 ,\n ==Inside load_module==\n );
ast_log(LOG_NOTICE ,\n ==Inside load_module==\n );
//tes4
//const char *data= (char*)malloc(sizeof(char) * 256);
char *sdata=4000,acPd;
ast_verb(3 ,\n ==Inside load_module sdata [%s] at [%p]
len[%d]\n ,sdata,sdata,strlen(sdata));
ast_log(LOG_NOTICE ,\n ==Inside load_module sdata [%s] at [%p]
and len[%d]\n ,sdata,sdata,strlen(sdata));
char *data= malloc(sizeof(char) * 256);
data=ast_strdupa(sdata);
ast_verb(3 ,\n ==Inside load_module data is [%s] at [%p]
len[%d]\n ,data,data,strlen(data));
ast_log(LOG_NOTICE ,\n ==Inside load_module data is [%s] at
[%p] and len[%d]\n ,data,data,strlen(data));
ast_verb(3 ,\n==Inside load_module data malloc == \n );
ast_log(LOG_NOTICE,\n==Inside load_module data malloc == \n
);
res |= ast_register_application_xml(app2,count_exec);
return res;
}
static int unload_module(void)
{
res |= ast_unregister_application(app2);
return res;
}
static struct ast_cli_entry cli_confbridge[] = {
AST_CLI_DEFINE(count_exec, MyConfbrigdeCount Show Number of
adminUser(s) in Conference. ),
}
static int count_exec(struct ast_channel *chan, const char *data)
{
int res = 0;
struct conference_bridge *conf=NULL;
int count;
char *localdata;
char val[80] = 0;
struct ao2_iterator i;
struct conference_bridge tmp;
AST_DECLARE_APP_ARGS(args,
AST_APP_ARG(confno);
AST_APP_ARG(varname);
);
ast_verb(3,\nInside count_exec =\n);
ast_verb(3,\n = 0xfffe inside count_exec == data[%s] at add
:[%p] ,len:[%d] \n,data,data,strlen(data));
return res;
}
===
(gdb) bt
#0 __strlen_sse2_bsf () at ../sysdeps/i386/i686/multiarch/strlen-sse2-
bsf.S:64
#1 0x00cefa49 in count_exec (chan=0xd09d78, data=0xfffe Address
0xfffe out of bounds) at app_confbridge.c:2438
#2 0x080d40eb in __ast_cli_register (e=0xd09d78, ed=0x0) at cli.c:2118
#3 0x080d4459 in ast_cli_register (e=0xd09d78) at cli.c:2178
#4 0x080d4482 in ast_cli_register_multiple (e=0xd09900, len=13) at
cli.c:2189
#5 0x00cf8030 in load_module () at app_confbridge.c:4779
#6 0x0812ba89 in start_resource (mod=0x905e740) at loader.c:845
#7 0x0812c45c in load_resource_list (load_order=0xbfdbb8b0,
global_symbols=0, mod_count=0xbfdbb8a8) at loader.c:1045
#8 0x0812ca5a in load_modules (preload_only=0) at loader.c:1198
#9 0x080895f7 in main (argc=4, argv=0xbfdbcdc4) at asterisk.c:4180
(gdb) frame 1
#1 0x00cefa49 in count_exec (chan=0xd09d78, data=0xfffe Address
0xfffe out of bounds) at app_confbridge.c:2438
2438ast_verb(3,\n = 0xfffe inside count_exec == data add
:%p ,len:%d \n,data,strlen(data));
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Re: [asterisk-users] Regarding SIP-T/SIP-I support in Asterisk.
Amit, I know how to play with SIP in asterisk and other tools . I want to know weather asterisk natively support or is there any extra patch or any workaround for SIP-T/SIP-I. Regarding packets and other things I am still not integrating it . I am searching some open-source tool which can send generate this type of packets and structure . Once I will integrate to our provider I will definitely check and share with experts here. On Thu, Mar 13, 2014 at 11:13 AM, Amit a...@avhan.com wrote: Hi Dhaval, If you capture and share SIP traces for inbound and outbound calls separately, experts on this list can guide to achieve objective. You can enable SIP trace on asterisk by executing following command in Asterisk console *sip set debug on* *Thanks Regards,* Amit Patkar On 3/12/2014 11:17 PM, DHAVAL INDRODIYA wrote: Thanks Amit, I want following scenario. INCOMINGCALL --- MSC (SIP-T) PBX (Asterisk) OUTGOINGCALL --- PBX (Asterisk) (SIP) to (SIP-T) --- Aircel MSC I understood that via Dial-plan we can achieve and get extra parameters values. But what about RTP fields as per my analysis ISUP packets are not sending RTP/AVP they are sending multipart data. please correct me if can achieve this functionality. Thanks Dhaval On Wed, Mar 12, 2014 at 6:15 PM, Amit a...@avhan.com wrote: Hi Dhaval, Theoretically, Asterisk can support SIP-I / SIP-T. Since protocols provide additional information and controls, you will not get those benefits. You will have to write dial plan functions to extract addition information exposed by SIP-I / SIP-T. Though, I have not tested it with Asterisk, I have successfully deployed application on other SIP platforms and interoperability with SIP-I/SIP-T was not an issue. *Regards,* Amit Patkar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] warnign
Vladimir Mikhelson vlad at mikhelson.com writes: Hi, Here is the reply from the developer as to what can be done immediately to remove the offending logging. He can just ignore these messages, they say that chan_ooh323 don't known indication signal 33 (AST_CONTROL_PVT_CAUSE_CODE) and processing of this signal isn't mandatory. If he would to remove this messages he can comment ast_log string in chan_ooh323 in ooh323_indicate function on the bottom of function: default: ast_log(LOG_WARNING, Don't know how to indicate condition %d on %s\n, condition, callToken); Hi! Thanks a lot for your post Have same problem with asterisk 11.8.1 + dongle. I Have this message - WARNING[8081][C-0007]: channel.c:1002 channel_indicate: [Dongle/gsm-mega-08-010007] Don't know how to indicate condition 33. So i find a file named channel.c and comment like that - /* default: ast_log (LOG_WARNING, [%s] Don't know how to indicate condition %d\n, ast_channel_name(channel), condition); res = -1; break; */ And it's work =) Warning is gone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to manage remote agents with Queue
Hello, I've got a small install with Asterisk 11. This box is connected to PSTN through a SIP trunk. I need to add a cellular phone as a remote agent of an existing queue. At the moment, this queue is configured according a ringall strategy and busy agent can be dialed. My plan is : 1- to create a remote-agent context and insert a Wait statement into it so that local agents would get dialed ahead of remote agents. 2- to avoid dialing remote agents already on call (with local phones) by hanging up calls in remote-agent context when conditions are met. 3- use local channels such as Local/123@remote-agent My questions relate to the above point 2. Though I don't need at the moment, which state interface can I use to tell Queue application a (SIP) remote agent is Busy or not ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange records in cdr
Hi, I have asterisk-server version of 1.8.11-cert7. When external enemy try to using it for calling, server write this string to log: 2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [9810972592309759@from-sip-external:1] NoOp(SIP/external-ip-00065fd2, Received incoming SIP connection from unknown peer to 9810972592309759) in new stack [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [9810972592309759@from-sip-external:2] Set(SIP/external-ip-00065fd2, DID=9810972592309759) in new stack [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [9810972592309759@from-sip-external:3] Goto(SIP/external-ip-00065fd2, s,1) in new stack [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Goto (from-sip-external,s,1) [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [s@from-sip-external:1] GotoIf(SIP/external-ip-00065fd2, 0?checklang:noanonymous) in new stack [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Goto (from-sip-external,s,5) [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [s@from-sip-external:5] Set(SIP/external-ip-00065fd2, TIMEOUT(absolute)=15) in new stack [2014-03-13 09:56:04] VERBOSE[4754] func_timeout.c: Channel will hangup at 2014-03-13 09:56:19.412 MSK. [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [s@from-sip-external:6] Answer(SIP/external-ip-00065fd2, ) in new stack [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: == Spawn extension (from-sip-external, s, 6) exited non-zero on 'SIP/external-ip-00065fd2' [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: -- Executing [h@from-sip-external:1] Hangup(SIP/external-ip-00065fd2, ) in new stack [2014-03-13 09:56:04] VERBOSE[4754] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/external-ip-00065fd2' It's correct, such call don't allowed , my money is safe. mysql select calldate, clid,src,dst,dcontext,channel,lastapp,duration,billsec,disposition,amaflags from cdr where calldate like '2014-03-13 09:56:04'; +-+-+-+-+---++-+--+-+-+--+ | calldate| clid| src | dst | dcontext | channel | lastapp | duration | billsec | disposition | amaflags | +-+-+-+-+---++-+--+-+-+--+ | 2014-03-13 09:56:04 | 100 100 | 100 | s | from-sip-external |SIP/ip-00065fd2 | Answer |0 | 0 | ANSWERED|3 | +-+-+-+-+---++-+--+-+-+--+ What is clid 100 100? Why it came from? No this source into log. -- Игорь Гайсин Email: igor.gaj...@tts.tv Телефон: 8-499-967-77-97 (4096) Должность: Системный администратор ООО Бриллианит -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
On Thu, Mar 13, 2014 at 4:18 AM, hkc323 hkc...@gmail.com wrote:
Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
function count_exec(struct ast_channel *chan, const char *data).But at
compile time char * data cause core dumped. Asterisk-11.5.1 Centos6
app_confbrige.c confbridge.conf
=
Task: Using Dailplan user want to retrive no of user in conference
'6050' = 1. Verbose(3,testMyConfbridgeCount) [pbx_config] 2.
MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,== ${count} )
[pbx_config]
Please discontinue spamming the users list with your posts.
Not receiving an answer to your question is not a reason to repeatedly
post (four posts now in the past few days?)
I've already responded to your original post and asking you to post on
the issue tracker and follow the issue guidelines to provide the
information needed to investigate the crash.
--
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200
Check us out at: http://digium.com http://asterisk.org
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Re: [asterisk-users] Remote extensions call drops after 20 seconds.
Thanks Steve. I think my problem is NAT. I'm using iptables, but I don't sure if I'm doing right steps. In the principal router I've forwarded the ports, but in my firewall (iptables on PBX server) I'm not sure. 201.237.180.154 is my remote place. #El NAT para el 5060 y el 1-3 (rtp) iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A POSTROUTING --proto udp --src 192.168.1.180 -j MASQUERADE iptables -t filter -A FORWARD --proto udp --dport 5060 -j ACCEPT iptables -t filter -A FORWARD --proto udp --dport 1:3 -j ACCEPT Can somebody help me to configure my NAT on iptables ? Maybe an example. Thank you again. On Mon, Mar 10, 2014 at 4:31 PM, Steve Totaro stot...@totarotechnologies.com wrote: Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.comwrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.com wrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
Re: [asterisk-users] Remote extensions call drops after 20 seconds.
Guys, but ALL MY INCOMING CALLS (in remote extensions) WORKS FINE. Should be a NAT issue? On Thu, Mar 13, 2014 at 8:43 AM, alp...@gmail.com alp...@gmail.com wrote: Thanks Steve. I think my problem is NAT. I'm using iptables, but I don't sure if I'm doing right steps. In the principal router I've forwarded the ports, but in my firewall (iptables on PBX server) I'm not sure. 201.237.180.154 is my remote place. #El NAT para el 5060 y el 1-3 (rtp) iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.154 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 5060 -j DNAT --to 192.168.1.180 iptables -t nat -A PREROUTING --dst 201.237.180.158 --proto udp --dport 1:3 -j DNAT --to 192.168.1.180 iptables -t nat -A POSTROUTING --proto udp --src 192.168.1.180 -j MASQUERADE iptables -t filter -A FORWARD --proto udp --dport 5060 -j ACCEPT iptables -t filter -A FORWARD --proto udp --dport 1:3 -j ACCEPT Can somebody help me to configure my NAT on iptables ? Maybe an example. Thank you again. On Mon, Mar 10, 2014 at 4:31 PM, Steve Totaro stot...@totarotechnologies.com wrote: Check here: http://www.freepbx.org/forum/freepbx/installation/incoming-calls-drop-30-seconds-on-fpbx-2-9-0-1-8-7-0 Thanks, Steve Totaro On Mon, Mar 10, 2014 at 4:43 PM, alp...@gmail.com alp...@gmail.comwrote: Yes, well, really is Elastix. Hmmm where I need to pt directmedia=no ? Thanks, On Mon, Mar 10, 2014 at 2:38 PM, Eric Wieling ewiel...@nyigc.comwrote: Try ulaw instead of g729, set directmedia=no I see you are using FreePBX. I cannot help further. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Monday, March 10, 2014 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: and...@telesip.net Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. Guys, hi. I have not solved the problem. Outgoing calls to remote extensions drops on 5-20 seconds. Incoming calls work perfectly. Hope you can help me. Attach updated logs: http://pastebin.com/HmKEDAUq Thanks, On Thu, Dec 19, 2013 at 8:48 AM, Eric Wieling ewiel...@nyigc.com wrote: See sip.conf.sample in the Asterisk tarball for documentation of valid settings. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of alp...@gmail.com Sent: Wednesday, December 18, 2013 9:30 PM To: and...@telesip.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Remote extensions call drops after 20 seconds. I set canreinvite=very in the remote extension, and now the call not drops. Valid solution? On Wed, Dec 18, 2013 at 6:38 PM, Andres and...@telesip.net wrote: On 12/18/13, 3:09 PM, alp...@gmail.com wrote: Hello. I have a problem with the configuration of a remote extensions. Calls are truncated at 20 seconds. I got my my NAT firewall properly configured. Here I attached my debug in CLI: http://pastebin.com/gh34E69f When the call is setup I see your Asterisk retransmitting the SIP/2.0 200 OK packet many times and getting no response. The other end needs to receive the packet and generate an ACK. You need to trace where that packet is going and figure out why it is not reaching its target, or if it is, then why is the ACK not making it back. Thats your problem. Thank you! -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr -- Technical Support http://www.cellroute.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Allan Porras http://allanPorras.com http://www.AllanPorras.com Google Plus: http://goo.gl/BRkbX Twitter: @alpocr http://twitter/alpocr --
[asterisk-users] Replying to Posts
(If you want to reply to this message, this is not where your reply goes) Please, for the benefit of anyone reading the archives in search of answers to a question, when replying to messages on this list, can everyone try to follow the natural flow of conversation? That is, position your reply *AFTER* the thing you are replying to, not before it. You may remove quoted material in order to keep the message size down, but please leave enough of it to preserve context. (If you want to reply to a point made in the preceding paragraph, this is where your reply goes) If you need to make a point-by-point argument, split up your reply -- inserting artificial paragraph breaks into the quoted material, if necessary -- so each section of your reply follows the point it is addressing. (If you want to reply to a point made in the preceding paragraph, or the message as a whole, this is where your reply goes) -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
-1 Prefer top posting. Easy to see if I want to scroll down to see if it is something interesting to me. I get a lot of e-mails each day and scrolling wastes too much time. But if you have a solution to a problem that I raise, please feel free to post it anywhere you like. On 13/03/2014 11:33 AM, A J Stiles wrote: (If you want to reply to this message, this is not where your reply goes) Please, for the benefit of anyone reading the archives in search of answers to a question, when replying to messages on this list, can everyone try to follow the natural flow of conversation? That is, position your reply *AFTER* the thing you are replying to, not before it. You may remove quoted material in order to keep the message size down, but please leave enough of it to preserve context. (If you want to reply to a point made in the preceding paragraph, this is where your reply goes) If you need to make a point-by-point argument, split up your reply -- inserting artificial paragraph breaks into the quoted material, if necessary -- so each section of your reply follows the point it is addressing. (If you want to reply to a point made in the preceding paragraph, or the message as a whole, this is where your reply goes) -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, March 13, 2014 1:39 PM To: rwhee...@artifact-software.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replying to Posts On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. If Digium does not like my top posting then they can remove me from the mailing list. Your battle is already lost unless Outlook is banned from the mailing list. This is an example of why I top post. Who wrote what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CONNECTEDLINE(name) ISDN problem
When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I see that CONNECTEDLINE(num) maps to the connected number information element and CONNECTEDLINE(name) to the display element. The pcap trace does actually contain my CONNECTEDLINE(name) plus a leading byte with a value of 0xB1 (or \261 in octal notation). This additional byte is part of the announced string length. Now I wonder, whether this byte is causing the trouble. Does anybody know what this leading byte is actually doing there? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling [ewiel...@nyigc.com] Sent: Thursday, March 13, 2014 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, March 13, 2014 1:39 PM To: rwhee...@artifact-software.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replying to Posts On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. If Digium does not like my top posting then they can remove me from the mailing list. Your battle is already lost unless Outlook is banned from the mailing list. This is an example of why I top post. Who wrote what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ditto, bottom posting is from the 90's. We've passed that era. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Thursday, March 13, 2014 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts _ From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling [ewiel...@nyigc.com] Sent: Thursday, March 13, 2014 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, March 13, 2014 1:39 PM To: rwhee...@artifact-software.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replying to Posts On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. If Digium does not like my top posting then they can remove me from the mailing list. Your battle is already lost unless Outlook is banned from the mailing list. This is an example of why I top post. Who wrote what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ditto, bottom posting is from the 90's. We've passed that era. _ Spam http://spamaway.npgco.com/canit/b.php?i=01LB6IUVim=74a497f24a89t=201403 13c=s Not spam http://spamaway.npgco.com/canit/b.php?i=01LB6IUVim=74a497f24a89t=201403 13c=n Forget previous vote http://spamaway.npgco.com/canit/b.php?i=01LB6IUVim=74a497f24a89t=201403 13c=f -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
After each line of text, please also dip the corner of your keyboard into your ink well to ensure your writing can been seen. Calling something natural because it used to be that way isn't always correct. -MD- P.S. Notice how little we see PS in posts...now that we can also edit our own posts before clicking send. Again, just because it used to be doesn't make it right... P.P.S. This sounds like a fun debate that will go nowhere... From: asterisk-users-boun...@lists.digium.com asterisk-users-boun...@lists.digium.com on behalf of Matt Hoskins matt.hosk...@npgco.com Sent: Thursday, March 13, 2014 2:46 PM To: Asterisk Users List Subject: Re: [asterisk-users] Replying to Posts From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Thursday, March 13, 2014 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] on behalf of Eric Wieling [ewiel...@nyigc.com] Sent: Thursday, March 13, 2014 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.commailto:rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts -Original Message- From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, March 13, 2014 1:39 PM To: rwhee...@artifact-software.commailto:rwhee...@artifact-software.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replying to Posts On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. If Digium does not like my top posting then they can remove me from the mailing list. Your battle is already lost unless Outlook is banned from the mailing list. This is an example of why I top post. Who wrote what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Ditto, bottom posting is from the 90's. We've passed that era. Spam http://spamaway.npgco.com/canit/b.php?i=01LB6IUVim=74a497f24a89t=20140313c=s Not spamhttp://spamaway.npgco.com/canit/b.php?i=01LB6IUVim=74a497f24a89t=20140313c=n Forget previous votehttp://spamaway.npgco.com/canit/b.php?i=01LB6IUVim=74a497f24a89t=20140313c=f -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Thursday, March 13, 2014 2:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; rwhee...@artifact-software.com Subject: Re: [asterisk-users] Replying to Posts -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Thursday, March 13, 2014 1:39 PM To: rwhee...@artifact-software.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Replying to Posts On Thu, 13 Mar 2014, Ron Wheeler wrote: -1 Prefer top posting. Your preferences are in conflict with the mailing list rules (http://www.asterisk.org/community/discuss), specifically #5. It has to be all one way or the other. This is an English language list. Thus, the natural expectation is top to bottom, left to right, answers follow questions. If Digium does not like my top posting then they can remove me from the mailing list. Your battle is already lost unless Outlook is banned from the mailing list. This is an example of why I top post. Who wrote what? Ditto, bottom posting is from the 90's. We've passed that era. I bottom posted when I didn't have to use a PoS e-mail client like Outlook. Much like the battle of sip trunks, the battle for top posting .vs. bottom posting is already lost. I must admit, this is more fun than I expected. Sort of a treasure hunt of who said what when in the message. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CONNECTEDLINE(name) ISDN problem
On Thu, Mar 13, 2014 at 1:42 PM, jg webaccou...@jgoettgens.de wrote: When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I see that CONNECTEDLINE(num) maps to the connected number information element and CONNECTEDLINE(name) to the display element. The pcap trace does actually contain my CONNECTEDLINE(name) plus a leading byte with a value of 0xB1 (or \261 in octal notation). This additional byte is part of the announced string length. Now I wonder, whether this byte is causing the trouble. Does anybody know what this leading byte is actually doing there? AFAIK, It is some kind of character set code. That byte is intentionally put there for switches that are not Q.SIG or ETSI. Technically, the display IE is only to be sent from the network to the user. So sending it to the network is undefined and switch dependent. Connected line name support is fully supported only by Q.SIG since it actually defines how to pass the name. Using the display IE is a defacto standard but is only really going to work in the network to user direction. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. On 13/3/14 5:13 pm, Ron Wheeler wrote: -1 Prefer top posting. Easy to see if I want to scroll down to see if it is something interesting to me. I get a lot of e-mails each day and scrolling wastes too much time. You can then also reply to another point here like this: it's as much about trimming previous posts as about not top posting. New posts should include just enough context to ensure the message isn't meaningless, but not quoting a 20+ line irrelevance. That way you see both the question and the answer without scrolling. Whilst we're on the subject of mailing lists, I'd like to add my personal pet rant: MTAs that don't add/honour In-Reply-To headers. Completely breaks threaded readers. That is all :-) Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CONNECTEDLINE(name) ISDN problem
AFAIK, It is some kind of character set code. That byte is intentionally put there for switches that are not Q.SIG or ETSI. Technically, the display IE is only to be sent from the network to the user. So sending it to the network is undefined and switch dependent. Got it. The single instance where I saw a string on the display was when I called my telco's technical support. Connected line name support is fully supported only by Q.SIG since it actually defines how to pass the name. Using the display IE is a defacto standard but is only really going to work in the network to user direction. Too bad, I was already dreaming of offering some customers a few nifty features, like greetings, email addresses for further contacts, booking confirmation numbers for hotels, etc... I shall still go through all the charset options to see whether there is any effect. Meanwhile I managed to get a pcap trace of a calling device and the display element is not at all present. I guess they are filtering it out. Thank you, Richard. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf
On Thu, Mar 13, 2014 at 9:24 AM, Rusty Newton rnew...@digium.com wrote:
On Thu, Mar 13, 2014 at 4:18 AM, hkc323 hkc...@gmail.com wrote:
Address 0xfffe out of bounds why and how to
solve.MyConfbridgeCount(conferencenumber,variablename )return total number
of user in conference given by conferencenumber otherwise zero.At runtime
using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call
function count_exec(struct ast_channel *chan, const char *data).But at
compile time char * data cause core dumped. Asterisk-11.5.1 Centos6
app_confbrige.c confbridge.conf
=
Task: Using Dailplan user want to retrive no of user in conference
'6050' = 1. Verbose(3,testMyConfbridgeCount) [pbx_config] 2.
MyConfbridgeCount(4000,count) [pbx_config] 3. verbose(3,== ${count} )
[pbx_config]
Please discontinue spamming the users list with your posts.
Not receiving an answer to your question is not a reason to repeatedly
post (four posts now in the past few days?)
I've already responded to your original post and asking you to post on
the issue tracker and follow the issue guidelines to provide the
information needed to investigate the crash.
Actually, in this case, he shouldn't post a bug report to the issue
tracker. The bug he is encountering is with some custom code in
ConfBridge, namely with the application MyConfbridgeCount:
static const char *const app2 =MyConfbridgeCount;
You should contact the author of that code and ask them to fix the crash.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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[asterisk-users] sorry for askingm but I can#r fund a solution
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register: Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching peer found I tried hard bit can's find a solution or even a hint ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorry for askingm but I can#r fund a solution
[Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register: Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching peer found What does the cli command sip show peers show? Do you have a definition for the sip device 2000 in sip.conf? jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] strange records in cdr
On Thu, Mar 13, 2014 at 7:58 AM, Игорь Гайсин igor.gaj...@tts.tv wrote: snip mysql select calldate, clid,src,dst,dcontext,channel,lastapp,duration,billsec,disposition,amaflags from cdr where calldate like '2014-03-13 09:56:04'; +-+-+-+-+---++-+--+-+-+--+ | calldate| clid| src | dst | dcontext | channel | lastapp | duration | billsec | disposition | amaflags | +-+-+-+-+---++-+--+-+-+--+ | 2014-03-13 09:56:04 | 100 100 | 100 | s | from-sip-external |SIP/ip-00065fd2 | Answer |0 | 0 | ANSWERED|3 | +-+-+-+-+---++-+--+-+-+--+ What is clid 100 100? Why it came from? No this source into log. That is the Caller ID information for that channel. -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorry for askingm but I can#r fund a solution
This mean that you either do not have such peer in your "sip.conf" or "permit" option of the peer 2000 don not include IP address 2000. On 03/13/2014 09:43 PM, q...@vienna.at wrote: [Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register: Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching peer found I tried hard bit can's find a solution or even a hint ru -- Mit freundlichen Gren / Best regards Vadim Lungu System Engineer Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com Web https://www.yopeso.com YOPESO s.r.l. Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On 3/13/2014 11:33 AM, A J Stiles wrote: If you need to make a point-by-point argument, split up your reply -- a critical piece to this component is proper quoting. the person replying needs to differentiate between what he is writing and what is is replying to. notice the in front of what I am quoting, above. in addition, clicking reply, quoting 100 lines, and then adding a 1 line response is lazy. trim the quotation to what makes sense. that said, i love a good top-post flame thread, so this should be interesting to watch. I'll start off by saying the biggest whine i hear is that my MUA doesn't support bottom-posting, which holds no water. i dont care that much, though- i don't waste time on top-posted messages a nor messages that are quoted stupidly. -- Jeremy Kister http://jeremy.kister.net./ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] func_odbc do not read LIKE predicate
Hello everyone,
I would be extremely glad if someone could help me with the
following issue:
cat /etc/asterisk/func_odbc.conf
[call_user]
prefix=GET
dsn=asterisk_odbc_sip
readsql=SELECT name FROM asterisk_sippeers WHERE name = '%477'
Asterisk CLI show the following issue:
-- Executing [205@phones_wildcard:1]
NoOp("SIP/Y_MD_vlungu_477-0008",
"000") in new stack
-- Executing [205@phones_wildcard:2]
NoOp("SIP/Y_MD_vlungu_477-0008", "205") in new stack
Found no rows [SELECT name FROM asterisk_sippeers WHERE
name = '%477']
-- Executing [205@phones_wildcard:3]
Set("SIP/Y_MD_vlungu_477-0008", "YUID=") in new stack
-- Executing [205@phones_wildcard:4]
NoOp("SIP/Y_MD_vlungu_477-0008",
"##") in new stack
-- Executing [205@phones_wildcard:5]
NoOp("SIP/Y_MD_vlungu_477-0008",
"##") in new stack
-- Executing [205@phones_wildcard:6]
Macro("SIP/Y_MD_vlungu_477-0008", "simple,,30") in new stack
Anyone has any ideas what is going wrong ?
--
Mit freundlichen Gren / Best regards
Vadim Lungu
System Engineer
Tel
+49-941-569592-0
Fax
+49-941-569592-99
Mail
vadim.lu...@yopeso.com
Web
https://www.yopeso.com
YOPESO s.r.l.
Calea Orheiului 20/1
2059 Chisinau
Republik Moldau
Registergericht Chisinau
Registernummer 1008600037623
VAT-ID
0207159
Geschftsfhrer Svetlana Arnaut
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Re: [asterisk-users] func_odbc do not read LIKE predicate
readsql=SELECT name FROM asterisk_sippeers WHERE name = '%477' Not sure what database you are accessing, but have you tried the following: readsql=SELECT name FROM asterisk_sippeers WHERE name LIKE '%477'-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorry for askingm but I can#r fund a solution
On Thu, Mar 13, 2014 at 08:49:32PM +0100, jg wrote: [Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register: Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching peer found What does the cli command sip show peers show? Not a single 2000, no 2000 at all. Do you have a definition for the sip device 2000 in sip.conf? No one. A lot of others but no 2000. ru -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc do not read LIKE predicate
Sure , here is the reasult. mysql SELECT name FROM asterisk_sippeers WHERE name LIKE '%477' ; +-+ | name | +-+ | Y_MD_vlungu_477 | +-+ 1 row in set (0.00 sec) On 03/13/2014 11:17 PM, Kevin Larsen wrote: SELECT name FROM asterisk_sippeers WHERE name LIKE '%477' -- Mit freundlichen Gren / Best regards Vadim Lungu System Engineer Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com Web https://www.yopeso.com YOPESO s.r.l. Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorry for askingm but I can#r fund a solution
On Thu, Mar 13, 2014 at 10:41:58PM +0200, Vadim Lungu wrote: This mean that you either do not have such peer in your sip.conf or permit option of the peer 2000 don not include IP address 2000. I neither have a 2000 in sip,conf nor I want to have one. 2000 doesn't have an IP and I want to get rid of it, honestly. I'd really want to know, where this 2000 is burned in and how to erase it. sip show peers does'nt show a peer 2000 nor I have a user 2000. On 03/13/2014 09:43 PM, q...@vienna.at wrote: [Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register: Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching peer found I tried hard bit can's find a solution or even a hint ru -- Mit freundlichen Grüßen / Best regards Vadim Lungu *System Engineer* Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com mailto:vadim.lu...@yopeso.com Web https://www.yopeso.com *YOPESO s.r.l.* Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschäftsführer Svetlana Arnaut -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorry for askingm but I can#r fund a solution
I neither have a 2000 in sip,conf nor I want to have one. 2000 doesn't have an IP and I want to get rid of it, honestly. I'd really want to know, where this 2000 is burned in and how to erase it. sip show peers does'nt show a peer 2000 nor I have a user 2000. Something that lives at 10.0.1.4 thinks it is extension 2000 and is trying to register. Your problem isn't Asterisk per se, it is finding where that IP address is located and what the device is to either fix its configuration to what it should be or to take the rogue device off your network.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc do not read LIKE predicate
Sure , here is the reasult. mysql SELECT name FROM asterisk_sippeers WHERE name LIKE '%477' ; +-+ | name| +-+ | Y_MD_vlungu_477 | +-+ 1 row in set (0.00 sec) What happens when you use that in your func_odbc.conf? Does your dialplan work?-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Kevin Larson sez: Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. I'll vote again for top posting, and expect my vote to be recognized internationally about as much as the Crimean referendum. --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
Don Kelly wrote: On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Kevin Larson sez: Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. I’ll vote again for top posting, and expect my vote to be recognized “internationally” about as much as the Crimean referendum. --Don As an interesting aside, the oft quoted rule #5 didn't exist for many years, until one of these diatribes took place. Then, and only then, was it added and the contention made that it was always there. Many of the same who continue to carp on top posting are the worst offenders when it comes to trimming the footers that arrive with each message, forcing the reader to wade through many of these to ( sometimes ) find a reply or maybe, just maybe, an answer. Often it isn't worth the effort to scroll through all the crap to find the pony. John Novack -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replying to Posts
On 13/03/2014 9:32 PM, Don Kelly wrote: On 13/3/14 6:27 pm, Eric Wieling wrote: This is an example of why I top post. Who wrote what? Of course, if you use a mail client that's capable of quoting correctly, it all works beautifully. Kevin Larson sez: Outlook can quote correctly, but it is an all or nothing setting it would appear. Lotus Notes actually handles it better as there is a Reply option for normal email and a Reply With Internet-Style History that I use for this list. I don't have any problems following the rules of the list, but I am fully on the side of the Replies should go at the top group and would vote for a change in the rules. I'll vote again for top posting, and expect my vote to be recognized internationally about as much as the Crimean referendum. But the Russians will get to keep Crimea so don't worry too much about our preference for top posting. In the long run. --Don -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] func_odbc do not read LIKE predicate
Vadim Lungu try this one . readsql=SELECT name FROM asterisk_sippeers WHERE name LIKE '%477'; -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sorry for askingm but I can#r fund a solution
I neither have a 2000 in sip,conf nor I want to have one. 2000 doesn't have an IP and I want to get rid of it, honestly. I'd really want to know, where this 2000 is burned in and how to erase it. This is IP is brute forcing you with peer that don't exist. So just block it. iptalble -A INPUT -s 10.0.1.4 -p udp -dport 5060 -j DROP On 03/14/2014 12:15 AM, q...@vienna.at wrote: On Thu, Mar 13, 2014 at 10:41:58PM +0200, Vadim Lungu wrote: This mean that you either do not have such peer in your "sip.conf" or "permit" option of the peer 2000 don not include IP address 2000. I neither have a 2000 in sip,conf nor I want to have one. 2000 doesn't have an IP and I want to get rid of it, honestly. I'd really want to know, where this 2000 is burned in and how to erase it. "sip show peers" does'nt show a peer 2000 nor I have a user 2000. On 03/13/2014 09:43 PM, q...@vienna.at wrote: [Mar 13 20:36:55] NOTICE[20590]: chan_sip.c:21812 handle_request_register: Registration from 'sip:2000@10.0.1.4' failed for '10.0.1.4' - No matching peer found I tried hard bit can's find a solution or even a hint ru -- Mit freundlichen Gren / Best regards Vadim Lungu *System Engineer* Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com mailto:vadim.lu...@yopeso.com Web https://www.yopeso.com *YOPESO s.r.l.* Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mit freundlichen Gren / Best regards Vadim Lungu System Engineer Tel +49-941-569592-0 Fax +49-941-569592-99 Mail vadim.lu...@yopeso.com Web https://www.yopeso.com YOPESO s.r.l. Calea Orheiului 20/1 2059 Chisinau Republik Moldau Registergericht Chisinau Registernummer 1008600037623 VAT-ID 0207159 Geschftsfhrer Svetlana Arnaut -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
