[asterisk-users] hangup locked channels
Hello, When I run "core show channels verbose", I seen around 10 locked channels that are lasting hundreds of hours (I haven't restarted Asterisk for more than 7 weeks). I try to hangup those channels using "hangup request ", but nothing happens. What could I do (besides restarting Asterisk)? Thanks Dov -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax and URA
Hi, I have an URA that says to my customers: "Dial 1 for support, dial 2 for sending a fax". This URA starts with g729, but when the call is transferred to the RxFax, it should be converted into g711, for the fax to work. Is there a way to solve this??? Thank you!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip channel variables
Hi, I am making a Dialplan in which I have to record some outgoing calls from some users and not from the others. I made to outgoing calls Macros, when that records calls (using Monitor()), and another that doesn't: [macro-outgoinglocal] and [macro-outgoinglocalrecord] Is it possible to have a user defined variable in sip.conf, so that before calling a Macro I can check the value of this variable and call the correct Macro? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Warning message
Is this message "normal"??? Apr 18 16:26:29 WARNING[1229]: channel.c:1323 ast_hangup: Hard hangup called by thread 51792816 on Local/[EMAIL PROTECTED],1ZOMBIE, while fd is blocked by thread 51792816 in procedure ast_waitfor_nandfds! Expect a failure RegardsDov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] log messages...
Hi, Gere are some messages that sometimes show up in my Asterisk logs... If you help me out to solve them, I could make a list of all know warning messagesso that we can publish in the wiki or somewhere else! - "res_features.c: Did not read data." - on Google, the only reference to this was in Russian :( - "Asked to transmit frame type 64, while native formats is 256 (read/write = 256/256)" - I am using codecs g711 (for fax only) and g729 - "channel.c: Avoided deadlock for '0x87421a8', 10 retries!" - "res_features.c: Don't know what to do about control frame: -1" - "rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 200.234.206.49" - debug: "chan_sip.c: That's odd... Got a response on a call we dont know about. Cseq 102 Cmd SIP/2.0" Besides that, afer the following log debug messages, at 13:54:59, my Asterisk went down Does the log below help any of you to help me? Apr 11 13:53:44 WARNING[5976] chan_sip.c: Asked to transmit frame type 64, while native formats is 256 (read/write = 256/256)Apr 11 13:53:44 NOTICE[5976] res_features.c: Don't know what to do about control frame: -1Apr 11 13:53:51 WARNING[5726] channel.c: Avoided deadlock for '0x88b84f8', 10 retries!Apr 11 13:53:57 WARNING[3826] res_features.c: Did not read data.Apr 11 13:53:59 WARNING[6089] res_features.c: Did not read data.Apr 11 13:54:02 NOTICE[29559] chan_sip.c: Peer 'gna_out_3060' is now UNREACHABLE! Last qualify: 30Apr 11 13:54:12 NOTICE[29559] chan_sip.c: Peer 'gna_out_3060' is now REACHABLE! (34ms / 1ms)Apr 11 13:54:13 WARNING[6089] res_features.c: Did not read data.Apr 11 13:54:13 WARNING[14471] channel.c: Thread 61332400 Blocking 'Local/[EMAIL PROTECTED],1', already blocked by thread 68828080 in procedure ast_waitfor_nandfdsApr 11 13:54:22 WARNING[5726] channel.c: Avoided deadlock for '0x882de60', 10 retries!Apr 11 13:54:23 WARNING[6089] res_features.c: Did not read data.Apr 11 13:54:36 WARNING[5726] channel.c: Avoided deadlock for '0xb72c9c58', 10 retries!Apr 11 13:54:59 NOTICE[6262] cdr.c: CDR simple logging enabled.Apr 11 13:54:59 DEBUG[6262] pbx_dundi.c: Seeding global EID '00:30:48:68:27:f4' from 'eth0'Apr 11 13:54:59 WARNING[6262] pbx.c: Requested contexts didn't get merged And yesterday I had a "Apr 10 18:00:14 WARNING[26794] channel.c: Hard hangup called by thread 88976304 on Local/[EMAIL PROTECTED],1ZOMBIE, while fd is blocked by thread 88976304 in procedure ast_waitfor_nandfds! Expect a failure" Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] core dump...
Hi, I checked core file generated at /tmp after a downtime, here is what I got... Is anybody able to interpret what did Asterisk went down??? Thank you Dov Loaded symbols for /usr/lib/libstdc++.so.5Reading symbols from /lib/libgcc_s.so.1...done.Loaded symbols for /lib/libgcc_s.so.1Reading symbols from /usr/lib/asterisk/modules/app_txfax.so...done.Loaded symbols for /usr/lib/asterisk/modules/app_txfax.so#0 0x00ccd239 in free () from /lib/tls/libc.so.6(gdb) bt#0 0x00ccd239 in free () from /lib/tls/libc.so.6#1 0x0805a62f in ast_frfree (fr=0xd91cd8) at frame.c:281#2 0x003692ac in ast_bridge_call (chan=0x9ea83d0, peer=0xb6901bd8, config=0xa138a38) at res_features.c:1442#3 0x00f431e3 in try_calling (qe=0x833c8e0, options=0x833c8e0 "\210\236\001\nfila", announceoverride=0x833ca5c "", url="" "", go_on=0xb6901bd8) at app_queue.c:2273#4 0x00f3d9b4 in queue_exec (chan=0x9ea83d0, data="" at app_queue.c:3009#5 0x0808dd5f in pbx_extension_helper (c=0x9ea83d0, con=0x39653028, context=0x9ea8520 "macro-filagrupo1", exten=0x9ea8614 "s", priority=7, label=0x0, callerid=0xa04de08 "Queue", action="" at pbx.c:545#6 0x0808c4b7 in ast_spawn_extension (c=0x39653028, context=0x39653028 Address 0x39653028 out of bounds, exten=0x39653028 Address 0x39653028 out of bounds, priority=962932776, callerid=0x39653028 Address 0x39653028 out of bounds) at pbx.c:2218#7 0x0053aa5d in macro_exec (chan=0x9ea83d0, data="" at app_macro.c:210#8 0x0808dd5f in pbx_extension_helper (c=0x9ea83d0, con=0x39653028, context=0x9ea8520 "macro-filagrupo1", exten=0x9ea8614 "s", priority=1, label=0x0, callerid=0x9dbd938 "Macro", action="" at pbx.c:545#9 0x0808e9d4 in __ast_pbx_run (c=0x9ea83d0) at pbx.c:2218#10 0x0808f6af in pbx_thread (data="" at pbx.c:2505#11 0x00c06dd8 in start_thread () from /lib/tls/libpthread.so.0#12 0x00d38d1a in clone () from /lib/tls/libc.so.6(gdb) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] call center running Asterisk - sound quality - critical!
Hi, I am using Asterisk fora call center on a Dual Xeon machine.. I currently have 109 active channels 53 active calls Every body is complaining about quality and cpu is around 80% idle. Is there any tuning I can do??? Besides that, Asterisk normally goes down once or twice per day... Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pause / unpausequeuemember
Hi,
I wanted to use the same extensions for Pausing and
UnPausing queue members.
Is that a variable that is set up with the agent
status (on call, available, not logged, paused) so that I could use it to make
some logic in this extension?
exten =
111,1,Set(AGENTEPARADESLOGAR=${$[AGENTBYCALLERID_${CALLERIDNUM}]})exten
= 111,2,PauseQueueMember(|Agent/${AGENTEPARADESLOGAR})exten =
111,3,Hangup
Or the only way out is to have different extensions
for pausing and unpausing?
Thank you
Dov
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[Asterisk-Users] voicemail context issue
Hi,
I know this has already been discussed here, but I
still have the problem even with 1.2.6:
When I call a mailbox in a context "company" is
doesn't play my busy message... It goes directly to the temp
message...
Am I doing something wrong?
== Everyone is busy/congested at this time
(1:0/1/0) -- Executing
NoOp("SIP/200.234.208.250-0840f548", "Voicemail de [EMAIL PROTECTED]") in new stack
-- Executing VoiceMail("SIP/200.234.208.250-0840f548", "[EMAIL PROTECTED]") in new stack --
Playing '/var/spool/asterisk/voicemail/bawm/87/temp' (language
'pt') -- Playing 'vm-intro' (language 'pt') ==
Spawn extension (macro-ramais_sip, s, 224) exited non-zero on
'SIP/200.234.208.250-0840f548' in macro
Here are the "show voicemail users for company"
results
Context Mbox
User
Zone
NewMsgcompany 87 Dov
Bigio
0
Thank you
Dov
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[Asterisk-Users] queue issue
Hi, I have several queues configured at my call center for different support levels. Today, something weird happened: - A client called queue 1 and was answered by an agent - The agent transferred the call to an user (not a queue), by dialing the atxtransfer (1) key defined in features.conf - The user transferred the client to another Queue, by using the second channel and the XFer key of her EyeBeam softphone) - The client entered this second queue and was answered correctly by an analyst from this second queue. But, when I ran "show queue secondqueue" or "show agents", even though the analyst is busy, she appear as available and the call is not registered in queue_log or anywhere else. She also can receive other calls from this queue, since she is not considered busy by the Queue application. Has anybody already realized this issue? Is this a bug or a misuse? Thank you!!!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] statechange_queue
Hi, Sometimes my Asterisk displays the following error message... Mar 31 12:53:04 WARNING[17170]: app_queue.c:519 statechange_queue: Failed to create update thread! Has anybody seen it before? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] after-queues
Hi, I have the following requirement.. after a customer is answered bya Queue, I want him to be redirected to another extensions, where an IVR would answer and ask for his opinion about the analyst who just solved his issue. Is there a way to redirect him automatically, or do I have to ask my agents to manually transfer the users to this IVR extension? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax behind ATA
Hi, I have installed a fax machine on a HT 486 ATA in my office, and it works perfectly, to send and receive faxes. When I install the same ATA on a fax machine at home (behind a NAT, in case it matters) faxes are received correctly, but I cannot send. Asterisk keeps showing a message "Unknown RTP codec 96 received" I am using g711u as the default codec. Does anyone have any idea or have already been through this problem? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr data
Hello,
I have an E1 and the possibility to use different
caller ids in this E1, so, before a Dial, I always have a SetCallerIDNum("User",
number).
When I check the CDR, the originator of the calls
appears to be this "number" I set in the caller id, but not the actual user that
originated the call.
Is there a way to set a callerid for the outgoing
call, but on cdr records to leave the originator id?
I know I could use the CDR user field, but I am
already using it for other purposes!
Thank you very much
Dov
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[Asterisk-Users] error messages on /var/log/asterisk/messages
Hi, I am using 1.2.3, and sometimes I can see the following message: Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:+ 1^Mar 2 08:42:42 WARNING[25937] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. Any ideas? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues tranfers
Hi, In features.conf I have defined "atxfer = 1" So, when a customer calls my support queue, and the agent from my support queue needs to transfer the customer to the billing queue, the agent dials 1, hears a "transfer" message and then dials the billing queue extensions. The agent enters a queue. At this point, he can hang up and leave the customer in the queue. But instead of this, I need my agent to be able to take the call back, so that he can tell the customer that there is a long queue and the billing departement will call him later. Is there a way to do this? I know that I the call was transferred to a user (not a queue), if the user hangs up the call goes back to the agent. The problem is that in the case of the queue the Queue application doesn't hangup up the call where there is a queue, so I need another key (probably the same atxfer "1" key) to do this. Is there a way to do this??? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] disallow, allow codes
Hi, On the general section of my sip.conf I had a disallow=all. Then I put disallow=all, allow=g729, allow=ulaw on my users. It didn't work until I removed the disallow=all from the header. I know disallow=all in the header is totally useless in this case (since I have it for every user), but anyway, is this the correct behavior? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chanspy instability
Hi, I had3 users spying on a call from the queue. On the exact time that the 4th user called the ChanSpy extension, Asterisk went down! Is there something wrong with ChanSpy??? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk error
Hi, I got this message on my Asterisk messages file and after it Asterisk went down... 2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:+ 1^2006-02-18 08:13:55 WARNING[3214] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source. Any ideas? a Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue_log analysis
Hi, I am running a call center based on Asterisk and building some statistics based on the queue_log file. I have some doubts about it that maybe you could help (actually, maybe these doubts are suggestions for enhancements!): 1st Scenario - Agent receives the call, and puts it on parking for somebody else to pick it up. Parking # 7000 (for attender transfer) 1140013998|1140013990.2524619|queue1|NONE|ENTERQUEUE||callerid1140014001|1140013990.2524619|queue1|Agent/5225|CONNECT|31140014016|1140013990.2524619|queue1|Agent/5225|COMPLETEAGENT|3|151140014016|1140013990.2524619|queue1|NONE|EXITWITHKEY||1 == Problems:Shouldn't a transfer to the parking extension (7000) be logged? I cannot track the call after it was transferred, would it be possible, via the unique call id, to log other events related to this call on this queue_log file, specially who picked up the call (whether it was picked or not), and how long did it take?What is the meaning of EXITWITHKEY in this scenario? 2nd Scenario - Agent receives the call, and transfers it to somedy else using # 1140014059|1140014051.2524641|queue1|NONE|ENTERQUEUE||callerid1140014062|1140014051.2524641|queue1|Agent/5225|CONNECT|31140014074|1140014051.2524641|queue1|Agent/5225|TRANSFER|203|default1140014074|1140014051.2524641|queue1|NONE|EXITWITHKEY||1 == Problems:I cannot track the call after it was transferred, would it be possible, via the unique call id, to log other events related to this call on this queue_log file, specially how long did it take?What is the meaning of EXITWITHKEY in this scenario? 3rd Scenario - Agent receives the call, and makes a blind transfer using the Transfer button of the phone (in my test, EyeBeam) 1140014104|1140014096.2524649|queue1|NONE|ENTERQUEUE||callerid1140014106|1140014096.2524649|queue1|Agent/5225|CONNECT|21140014129|1140014096.2524649|queue1|Agent/5225|TRANSFER|203|default == Problems:I cannot track the call after it was transferred, would it be possible, via the unique call id, to log other events related to this call on this queue_log file, specially how long did it take? 4th Scenario - Agent receives the call, and makes an attended transfer (putting the call on hold, dialing via another channel, andusing the Transfer button of the phone (in my test, EyeBeam) 1140014161|1140014153.2524663|queue1|NONE|ENTERQUEUE||callerid1140014164|1140014153.2524663|queue1|Agent/5225|CONNECT|31140014203|1140014153.2524663|queue1|Agent/5225|COMPLETEAGENT|3|39 == Problems: No transfer information is logged. Agent is considered busy (on call) until the call is actually ended, independent of the moment he actually transferred. In my agents opinion, the best way to make transfers would be the 3rd and 4th scenarios, which are obvious for phone users. But for their managers, scenarios 1 and 2 are better since more information can be used for their daily statistics. Anyway, even scenarios 1 and 2 miss lack some important statistics. Is there anybody working on enhancing this queue_log features or using any other way (maybe events and AMI) to make more complete statistic reports of call centers? Thank you very much Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Developing a call centre app. Communicationwithasterisk?
For java based applications, I'd recommend http://www.asteriskjava.org/latest/ - Original Message - From: yusuf [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 14, 2006 10:34 AM Subject: Re: [Asterisk-Users] Developing a call centre app. Communicationwithasterisk? Arne Morten Johansen wrote: Hi there. We're going to develop a call centre app for internal use in our office. The call centre is probably going to be a java-based client installed on a windows machine that our secretary can use. Features should be a way to see incoming calls, answer them, and then transfer the calls to our different users/groups/divisions. If it also could be possible to have a way to see if the user is registered, busy, unavailable or available etc before she makes the transfer would be great. We have some people that are very good at programming. But for them to go on, I need to layout a plan for them on how to communicate with the Asterisk server. They have no experience with Asterisk at all, and I'm not a good programmer. My first thought is calling a PHP-script from asterisk that communicates with the java-client through IP-sockets. But I don't see how this can make the applet able to transfer calls. I'm really stuck. Anyone got suggestions and tips? Any help would be greatly appreciated. Hi Arne, you would do this using the asterisk manager interface. Read up about it. there is a manager.conf file, where u set up username/passwd. There is java classes that talk to the manager interface, that can pick up any call events, which will allow you to pick up, transfer , answer. yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] about g729 license
Hi, I know that this topic has already been posted to this list previously, but each time the list grows bigger it is more difficult to find things.. Sorry to post this again then! Does the message below mean that I would need 15+36 licenses? lv09*CLI show g72915/36 encoders/decoders of 50 licensed channels are currently in use Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] about g729 license
Got it.. so, in this case, I am using 36 licenses, right?? Thank you very much Dov - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 14, 2006 1:25 PM Subject: Re: [Asterisk-Users] about g729 license Dov Bigio wrote: Does the message below mean that I would need 15+36 licenses? lv09*CLI show g729 15/36 encoders/decoders of 50 licensed channels are currently in use No. If you did, then you would have run out already, since you only have 50. Each license gets you one encoder and one decoder; use of either one consumes that license, but use of both still uses only one license. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr (again) and deadlocks
Hello, Today I had again problems with CDR. My MySQL cdr table was corrupted and thus CDR couldn't be logged. At this moment Asterisk console started to display the following message "Avoided deadlock for '0x843fa98', 10 retries!" hundreds, thousands of times (together with the table corrupted message), until it simply displayed a "Terminated" message and went down. I had to repair the MySQL table, and then restart Asterisk. The table corrupted message was useful for me to identify the corrupted table and repair it... but wouldn't it be possible that Asterisk would not "Terminate" because of this? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk logger - urgent!!!
Hi, Since yesterday my Asterisk 1.2.3 is displaying the following message every few seconds Asterisk Event Logger restartedRotated Logs Per SIGXFSZ (Exceeded file size limit) This causes my log files (verbose, queue_log) to become huge with lots of logger rotate messages, but I don't know which files is exceeding size limit, since even if I delete all log files I still get this message. Any way, I have plenty of disk space and couldn't find the reason for this message. Please help me identify the issue!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asterisk logger - urgent!!!
I found the problem. Master.csv reached 2.0GB and since the moment this happened Asterisk went crazy! Since I am using cdr-mysql, how do I disable the use of csvs? Thank you Dov - Original Message - From: Dov Bigio To: asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 10:56 AM Subject: asterisk logger - urgent!!! Hi, Since yesterday my Asterisk 1.2.3 is displaying the following message every few seconds Asterisk Event Logger restartedRotated Logs Per SIGXFSZ (Exceeded file size limit) This causes my log files (verbose, queue_log) to become huge with lots of logger rotate messages, but I don't know which files is exceeding size limit, since even if I delete all log files I still get this message. Any way, I have plenty of disk space and couldn't find the reason for this message. Please help me identify the issue!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk logger - urgent!!!
Hi Tzafrir, The problem was the file Master.csv that had reached 2.0GB. I am writing a cron script to backup this file periodically and prevent this from happening. Any way, if any developers are reading this, I don't think that rotating asterisk logs is the best way to handle this problem! Maybe a more user-friendly message could be logged, infoming which file reached the 2.0GB. About your question, yes I do, for log files. Is logger rotate could also after I delete csv files? Thank you Dov - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 11:42 AM Subject: Re: [Asterisk-Users] asterisk logger - urgent!!! On Thu, Feb 09, 2006 at 10:56:18AM -0200, Dov Bigio wrote: Hi, Since yesterday my Asterisk 1.2.3 is displaying the following message every few seconds Asterisk Event Logger restarted Rotated Logs Per SIGXFSZ (Exceeded file size limit) This causes my log files (verbose, queue_log) to become huge with lots of logger rotate messages, but I don't know which files is exceeding size limit, since even if I delete all log files I still get this message. Unrelated to the origin of the problem: Do you run 'logger reload' after deleting those logs? Otherwise Asterisk still writes to the old (deleted) logs Any way, I have plenty of disk space and couldn't find the reason for this message. Please help me identify the issue! Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk logger - urgent!!!
Hi Kevin, I see... That's why you rotate asterisk logs everytime this message occurs.. it makes sense. Unfortunately, in my case, it was the CDR CSV files tha reached that size, so rotating logs was just worsening my situation, since asterisk started to generate rotated log files every few seconds because of that. Is there a way to rotate CDR CSV files via Asterisk, or should I handle this outside Asterisk? Thanks! Dov - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 09, 2006 2:01 PM Subject: Re: [Asterisk-Users] asterisk logger - urgent!!! Dov Bigio wrote: Any way, if any developers are reading this, I don't think that rotating asterisk logs is the best way to handle this problem! Maybe a more user-friendly message could be logged, infoming which file reached the 2.0GB. Unfortunately when we receive SIGFSZ from the kernel, we have no way to know which file caused it. The assumption in Asterisk is that the only files we write to that will ever reach that size are log files. If any other file does, there will be trouble, as you have seen. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IVR Menu
Hi,
I made a simple menu using the Background
application and some wav files. I converted the wav files using
forain*.wav;dosox"$a"-r8000-c1"`echo$a|sed-es/wav//`gsm";done
(from http://www.voip-info.org/wiki/index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk)
The first two files "01/bemvindo" and "01/menu_top"
are good. But the third file (01/menu_top), fails in the end of the sentence,
andthis message"Auto fallthrough, channel 'SIP/dov.bigio-ae4a'
status is 'UNKNOWN'" appears in the console.
-- Executing
Goto("SIP/dov.bigio-ae4a", "01.menu.locaweb|s|1") in new
stack -- Goto (01.menu.locaweb,s,1)
-- Executing Answer("SIP/dov.bigio-ae4a", "") in new stack
-- Executing SetMusicOnHold("SIP/dov.bigio-ae4a", "fila") in new
stack -- Executing Set("SIP/dov.bigio-ae4a",
"TIMEOUT(digit)=15") in new stack -- Digit timeout set to
15 -- Executing Set("SIP/dov.bigio-ae4a",
"TIMEOUT(response)=15") in new stack -- Response timeout
set to 15 -- Executing BackGround("SIP/dov.bigio-ae4a",
"01/bemvindo") in new stack -- Playing '01/bemvindo'
(language 'pt') -- Executing
BackGround("SIP/dov.bigio-ae4a", "01/menu_top") in new
stack -- Playing '01/menu_top' (language 'pt')
== Auto fallthrough, channel 'SIP/dov.bigio-ae4a' status is
'UNKNOWN'
Can anybody help me?
Thank you
Dov
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Re: [Asterisk-Users] limit sip sessions
I think I have the same issue... In case usershave an IP Phone on their desks and Softphones on their PCs and are configured with the same username extensions, which phone will ring? The one that last sent the REGISTER... This can be conflicting... - Original Message - From: Script Head To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 03, 2006 1:10 PM Subject: Re: [Asterisk-Users] limit sip sessions You should create a secret dialing prefix like if you wanted to dial 1555333222 the user would actually have to dial 548261555333222. This way, even if they snatch the username/password but do not know the prefix, they won't be able to dial. On 2/2/06, Miguel [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote:Shouldn't all sip users have different usernames?(or am I missing some vital detail here?)PaulHYes Paul, Im in El Salvador and my users like to "share" their usernames/passwords and the original owner doesnt like to pay for callshe hasnt made.---Miguel___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error cdr mysql addon
Hi, After installing mysql, mysql-devel mysql cdr add on, I get the following error when I start Asterisk: [res_config_mysql.so]2006-02-03 18:41:16 WARNING[24786]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: _intel_fast_memcpy My server has the following MySQL rpms: rpm -qa | grep MySQLMySQL-server-4.0.20-0MySQL-shared-compat-4.0.18-0MySQL-devel-4.0.20-0perl-DBD-MySQL-2.1021-3MySQL-client-4.0.20-0 Any ideas? Thank you!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callerid issue
Hello, In my sip.conf I have each IP phones defined as follows. [ext13]type=friendsecret=123qualify=yes[EMAIL PROTECTED]language=ptcontext=geralfromuser=ext13username=ext13host=dynamicdisallow=allallow=g729allow=ulaw But, when I call from ext13 to ext12, the caller id that appears on Phone12 is ext12, and not ext13, so when the users wants to dial to a missed call number, his phone simply calls to himself, and not the the right caller. I tried to change parameters callerid and fromuser, with no success. Even tried in extensions.conf to use SetCallerId, but nothing helped. Am I missing something, or isthere something wrong here? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (Un)PauseQeueMamber usage
Hi BJ,
I am trying your example, but I am getting calls to 'h' logged on CDR.
If I put NoCDR() on the h extension, priority 1, CDR stores my calls, but
with zero billable length.
I am really confused with h extension, NoCDR and ForkCDR ;)
Thank you
Dov
- Original Message -
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday, January 28, 2006 10:59 AM
Subject: Re: [Asterisk-Users] (Un)PauseQeueMamber usage
On 1/28/06, Joe [EMAIL PROTECTED] wrote:
Does anyone have an example of hoe to use these two commands? I have read
he
documentation, and I am still unclear on where this command goes, as part
of
extensions.conf or where?
If someone could post a working example it would be most helpful.
Here's how I've done it before for other clients:
On the dialout portion I've changed the dial plan to:
exten =
_1NXXNXX,1,GotoIf($[${LEN(${$[AGENTBYCALLERID_${CALLERID(num)}]})}
2]?2:3)
exten =
_1NXXNXX,2,PauseQueueMember(|Agent/${$[AGENTBYCALLERID_${CALLERID(num)}]
})
exten = _1NXXNXX,3,Dial(SIP/SIP PEER/${EXTEN},,Tg)
exten = _1NXXNXX,4,ForkCDR()
What that's basically saying is that if the calling number is also
logged in as an agent, go ahead and pause that queue member in all
queues that they belong to and then make the call. I'm doing the
GotoIf because there are other extensions in that same context that
may not be logged in as agents and I don't want to make that pqm call
(though there's no real harm in doing so, it'll just tell you there's
no Interface as specified) with.
Then, in that same context, you put the following in the h extension
exten = h,1,ForkCDR()
exten = h,2,GotoIf($[${LEN(${$[AGENTBYCALLERID_${CALLERID(num)}]})}
2]?3:4)
exten =
h,3,UnPauseQueueMember(|Agent/${$[AGENTBYCALLERID_${CALLERID(num)}]})
exten = h,4,NoOp(Done!)
ForkCDR is important because if you don't do it you're going to find
that the original CDR that used to contain the destination number in
it, now contains only the 'h' extension in it. You could also use
ResetCDR(w) here. Your choice really. ForkCDR will fork the one CDR
into two preserving the original dial information, and then you may
choose to do a NoCDR() or just deal with the additional CDR generated
to the 'h' extension by ignoring it when you parse CDRs.
Hope this helps.
--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[Asterisk-Users] moh clock
Hi, I had a wct1xxp in my asterisk server, but I migrated to a cisco sip gateway, and then unplugged the e1. I then changed zaptel's Makefile to include ztdummy and ran modprobe ztdummy Music on hold for queues is not working well... it is simply mute. I realized that, while waiting on a Queue, if I ran a reload, the music on hold starts being played for a few seconds and then stops, until I reload again. I am using 1.2.3, but this happens to me since 1.2.0 (it worked well on 1.0.10). When I ran lsmod, I see usb-uhci 26860 0 [ztdummy]zaptel 183680 78 [ztdummy wcusb wct1xxp] Does this make sense? Should I recompile zaptel? How do I remove wct1xxp? (the card is actually there, but it has no E1 in it anymore. Thank you very much Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] h extension
Hi,
I want to count the number of open Zap channels on
my server.
[outgoingzap]
exten = _0NXX,1,NoOp(Outgoing Local - 7
digs - ${EXTEN:1})exten = _0NXX,2,Set(ZAP01=$[${ZAP01} +
1]|g)exten = _0NXX,3,Set(UPDATED=true)exten =
_0NXX,4,Dial(${TRUNK}/${EXTEN},60)exten = _0NXX,6,Busyexten
= _0NXX,7,Playback(thank-you)include =
hangupcontext
[hangupcontext]exten = h,1,NoCDR()exten
= h,2,GotoIf($["${UPDATED}" != "true"]?5)exten =
h,3,Set(UPDATED="")exten = h,4,Set(ZAP01=$[${ZAP01} - 1]|g)exten
= h,5,Hangup
I am not sure about how to use NoCDR, ForkCDR and
ResetCDR.
If don't use any of them, every call generates an
extra CDR with dst = 'h'.
If I use NoCDR, the whole call is not
logged.
If I use ResetCDR, the call is logged with duration
= 0.
How should I implement this?
Thank you
Dov
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[Asterisk-Users] Fw: chanspy
Hi, I was only able to ChanSpy Agent channels. How do I monitor outgoing calls? Thank youDov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk down because of cdr
Ok.. but I don't use Real Time at all. I just use cdr_mysql. It would be smarter if it simply ignored MySQL outages or at least just logged, but without stopping. Regards dov - Original Message - From: Dovid Bender [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Cc: asterisk-users@lists.digium.com Sent: Tuesday, January 17, 2006 2:43 PM Subject: Re: [Asterisk-Users] asterisk down because of cdr When using asterisk real time, every time somehting occurs in asterisk it goes to the DB. If the DB isnt up natrually it dosent know what to do. So yes this behavior is perfectly normal. Dovid (Sorry about the spelling mistakes) --- Dov Bigio [EMAIL PROTECTED] wrote: Hello, After 2 weeks and 4 days without a problem, Asterisk went down. What happened is that I am using Asterisk 1.2.1 on a machine and have a MySQL for CDR on another machine. The machine with MySQL went down and the Asterisk box was unable to connect to MySQL. This made Asterisk to go down and it was unable to restart until MySQL was back. I know that Asterisk displays a lot of warnings, but still works, when the cdr table is corrupt. But isn't it a strange behaviour to go down when MySQL is down? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] red alarm?
Hi, What is the meaning of: Jan 17 18:05:21 WARNING[2388]: chan_zap.c:6315 handle_init_event: Detected alarm on channel 2: Red AlarmJan 17 18:05:21 WARNING[2388]: chan_zap.c:1432 zt_disable_ec: Unable to disable echo cancellation on channel 2Jan 17 18:05:21 WARNING[2388]: chan_zap.c:6315 handle_init_event: Detected alarm on channel 3: Red AlarmJan 17 18:05:21 WARNING[2388]: chan_zap.c:1432 zt_disable_ec: Unable to disable echo cancellation on channel 3 This happened once today with my 30 channels, but then everything came backto normal. Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk down because of cdr
Hello, After 2 weeks and 4 days without a problem, Asterisk went down. What happened is that I am using Asterisk 1.2.1 on a machine and have a MySQL for CDR on another machine. The machine with MySQL went down and the Asterisk box was unable to connect to MySQL. This made Asterisk to go down and it was unable to restart until MySQL was back. I know that Asterisk displays a lot of warnings, but still works, when the cdr table is corrupt. But isn't it a strange behaviour to go down when MySQL is down? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cmd Dial parameters
Hi, For the dial application, parameter g is described as g: When the called party hangs up, exit to execute more commands in the current context. I want the following priority (or at least a priority I can jump to) to be executed anyway, it doesn't matter which party hang up. Is there a way to do so? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queus agents
Hi all, I have agents who are members of more than one queue. When an agent is busy with queue A, he is not considered busy by queue B, and receives call (since his EyeBeam Softphone has 6 channels). Besides that, I use a monitoring tool that connects through the manager interfaces and run "show queues" and "show agents" to know agents statuses. I need Asterisk to consider the agent busy for both Queues when he is actually answering any queue. Is there a way to do this? It could even be a solution that would Pause the agent on the second queue while he is busy with the first (is there a way to do this inside the dialplan?).. I wouldn't link to have to do an external application to listen to events and pause the agents outside Asterisk... Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail
Hi, I have my voicemails accounts configured with delete=yes|attach=yes Today I had problems with my smtp server and messages were not sent to users, BUT were deleted from the server. Is there a way to delete voicemail msgs only if e-mail is sent successfully??? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queus agents
No, I am using different Agent IDs, since I need agents to answer just one queue at a time... that is, I need person A to answer to the sales queue today, and the support queue tomorrow, and maybe both queues at the same time on the day after... :( So it seems that there is not easy solution for me... I'll try with an external application using manager api to Pause agents on one queue when they are busy on the other Thank you DOv - Original Message - From: Johann [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Sent: Friday, January 13, 2006 6:38 PM Subject: Re: [Asterisk-Users] queus agents Dov Bigio wrote: Hi all, I have agents who are members of more than one queue. When an agent is busy with queue A, he is not considered busy by queue B, and receives call (since his EyeBeam Softphone has 6 channels). Are you using the same AgentID for the person being on both queue A and queue B? Besides that, I use a monitoring tool that connects through the manager interfaces and run show queues and show agents to know agents statuses. I need Asterisk to consider the agent busy for both Queues when he is actually answering any queue. Is there a way to do this? If your users have more than one AgentID they will get a call for each of those AgentIDs. There is a slight side affect to this however, if you are using callback agents. Then the user is automatically marked as available in both queues or logged off in both(and also on a call if either queue sends them a call). Agents and Queues only care about the AgentID...if multiple AgentIDs go to the same place the queue/agent system does not check nor care. It could even be a solution that would Pause the agent on the second queue while he is busy with the first (is there a way to do this inside the dialplan?).. I wouldn't link to have to do an external application to listen to events and pause the agents outside Asterisk... Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pattern mach doubt
Hi ALL, Is it possible to use symbols # and * in the dialplan for pattern matching? I am doing a "follow me" dial plan, and wanted that my users could dial everything in one shot. But, exten = 888*XXX*XXX,1,NoOp(Follow Me from XXX to XXX) doesn't seem to work... Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk initialization
That's great... I didn't know about the persistentagents features! I'll test it asap! Thank you Dov - Original Message - From: Alexander Lopez To: Dov Bigio ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, January 07, 2006 5:16 PM Subject: RE: [Asterisk-Users] Asterisk initialization Do not know what version you are running, But there are a few ways to do this. There is a persistant setting: from agents.conf ;; Define whether callbacklogins should be stored in astdb for; persistence. Persistent logins will be reloaded after; Asterisk restarts.;persistentagents=yes If you want to handle it outside of Asterisk via an AGI you can have your AGI execute: AgentCallbackLogin([AgentNo][|[options][|[EMAIL PROTECTED]): this is providing that you have the information saved in your DB. Personal Opinion: Use the builtin features with the persistentagents options and use the php script in the contribs directory to see who is on. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov BigioSent: Friday, January 06, 2006 4:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk initialization Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to generate a .call file that calls and extension that would call the AGI to log all the agents back on. Is there another way of running an AGI on initialization? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ChanSpy via external application
Hello, It didn't work... I used "Data: SIP/dov.bigio-9949" which was the channel being used, and the call I received just had beeps... no conversation. According to the documentation on (http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy), ChanSpy doesn't take a channel as parameter, does it? Thank you very much!! Dov - Original Message - From: Giovanni Miano To: Dov Bigio ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 05, 2006 7:01 PM Subject: Re: [Asterisk-Users] ChanSpy via external application Use channel of your agentChannel: SIP/dov.bigioMaxRetries: 3RetryTime: 40WaitTime: 25Context: 01.telecomApplication: ChanSpyData: SIP/234-ssnfPriority: 1Cheers,Giovanni Miano 2006/1/5, Dov Bigio [EMAIL PROTECTED]: Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call between his extension and the extension I have for ChanSpy, passing as parameter the Agent number. For testing this, I tried a call file on /var/spool/asterisk/outgoing Channel: SIP/dov.bigio --- This is meMaxRetries: 3RetryTime: 40WaitTime: 25Context: 01.telecomApplication: ChanSpyData: Agent/5450- This is the Agent I want to monitorPriority: 1 The problem is that ChanSpy doesn't accept "Agent/" as parameter, just "Agent". Is there a way to ChanSpy a specific know Agent? (Or at least to send via dtmf the Agent Number I want to monitor right after the ChanSpy application is called? Thank you very much!Dov ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk initialization
Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to generate a .call file that calls and extension that would call the AGI to log all the agents back on. Is there another way of running an AGI on initialization? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ChanSpy via external application
Hi, I have developped an application that monitors the status of my queues through the events triggered on the Manager Interface. This way, I can know the status of my Agent real time. Now, I have a new requirement that I must allow a manager to click on the Agent he wants to monitor and be able to monitor the call. My idea was to, when the user clicks on the Agent, I would Originate a call between his extension and the extension I have for ChanSpy, passing as parameter the Agent number. For testing this, I tried a call file on /var/spool/asterisk/outgoing Channel: SIP/dov.bigio --- This is meMaxRetries: 3RetryTime: 40WaitTime: 25Context: 01.telecomApplication: ChanSpyData: Agent/5450- This is the Agent I want to monitorPriority: 1 The problem is that ChanSpy doesn't accept "Agent/" as parameter, just "Agent". Is there a way to ChanSpy a specific know Agent? (Or at least to send via dtmf the Agent Number I want to monitor right after the ChanSpy application is called? Thank you very much!Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue features
Hi, I am using the Queue application for 5 queues I have in my Call Center, and will by the end of January, implement it for the rest of the company (another 10 queues). One of the main problems I face and my call center managers are worried about is the fact that when an agent uses the DND button of the Softphone, call center managers have no way of monitoring this. Is there a way to track this? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue features
But a peer whose Softphone is on DND mode is still considered available, isn't it? - Original Message - From: Giovanni Miano To: Dov Bigio ; Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, December 30, 2005 12:21 PM Subject: Re: [Asterisk-Users] Queue features You can check status of Peer with Asterisk Management Interface (AMI)www.voip-info.org/wiki-Asterisk+manager+APICheers,Giovanni Miano 2005/12/30, Dov Bigio [EMAIL PROTECTED]: Hi, I am using the Queue application for 5 queues I have in my Call Center, and will by the end of January, implement it for the rest of the company (another 10 queues). One of the main problems I face and my call center managers are worried about is the fact that when an agent uses the DND button of the Softphone, call center managers have no way of monitoring this. Is there a way to track this? Thank you Dov___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp fax
If you check the AsteriskGuru.com tutorial about this, he explains how to edit this files manually.. it is really simple! - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, December 29, 2005 5:59 PM Subject: RE: [Asterisk-Users] spandsp fax Do I need to compile first the app_rxfax.c app_txfax.c to get the .so files? If the answer is yes, how I do that command, just I'm not and expert on GCC. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Burton Sent: Thursday, December 29, 2005 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax Make it manually, because there is somme diff from 1.0.9 edit Makefile and add : everything after + Pierre Carlos Alperin wrote: Ok, Everything was fine up to the moment to run patch apps_makefile.patch Then I got Hunk 1 of 2, on the line 98 of the Makefile. This is the Makefile.rej output. As you can see, the line 98 includes some + signs that are in the apps_makefile.patch. [EMAIL PROTECTED] apps]# cat Makefile.rej ** 94,103 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq --- 98,113 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq However there is no request to take those lines of that file. Carlos Alperin [EMAIL PROTECTED] *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rehan Ahmed *Sent:* Wednesday, December 28, 2005 7:43 PM *To:* Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, *Dov Bigio* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; Asterisk Users
Re: [Asterisk-Users] agent logs
Have a look at /var/log/asterisk/queue_log It has to be enabled on logger.conf (queue_log=yes on the [genera] section). - Original Message - From: Hall, Eric M. To: asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 6:25 PM Subject: [Asterisk-Users] agent logs I'm looking for a ay to track when an agent logs inand logs out. Best if it could be put in a mysql db but a text file will be ok for now.. Any help would be great ! Thanks ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp fax
I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] recording queue calls
It helped, a lot!
Thank you
Dov
- Original Message -
From: Faris Raouf [EMAIL PROTECTED]
To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-CommercialDiscussion asterisk-users@lists.digium.com
Sent: Saturday, December 24, 2005 4:17 PM
Subject: Re: [Asterisk-Users] recording queue calls
Dov Bigio wrote:
Hi,
When I set monitor-format=wav49 on file queues.conf for a queue,
Asterisk records calls at /var/spool/asterisk/monitor. But the file
names it users are the call-ids of the calls.
Is there a way to change that, and use information such as date, time,
agent and queue to build the filename?
It would make the localization of such files much more easy.
Other useful that I miss is the capability to to allow the files to be
stored in different directories, such as
/var/spool/asterisk/monitor/queue1, /var/spool/asterisk/monitor/queue2,
and so on, based on the queuename. Is this possible by any means?
Hi,
Yes. All you need to do is use the following in your extension.conf at
the point before you call the queue
SetVar(MONITOR_FILENAME=foo)
or, if you are using 1.2.x
Set(MONITOR_FILENAME=foo)
For example, I have:
Set(MONITOR_FILENAME=${TIMESTAMP}-${CALLERIDNUM}-to-SALES-${UNIQUEID})
and then a little later on:
Queue(salesqueue|t|||60)
in my extensions.conf
Which sets the monitor filename to start with a timestamp, then the CID
of the caller, then the to-SALES is what I use to differentiate
between queues (I'd have a different Set command for a different queue).
I then add the UNIQUEID as a just in case to make absolutely sure
there's no way I'd ever have two files of the same name.
I hope this helps,
Faris.
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[Asterisk-Users] spandsp fax
Hi, I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.htmlto install faxing capability on my server. I get the following error messages... Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found[app_rxfax.so]Dec 27 12:14:27 WARNING[14679]: loader.c:334 __load_resource: No load_module in module /usr/lib/asterisk/modules/app_rxfax.soDec 27 12:14:27 WARNING[14679]: loader.c:341 __load_resource: No unload_module in module /usr/lib/asterisk/modules/app_rxfax.soDec 27 12:14:27 WARNING[14679]: loader.c:348 __load_resource: No usecount in module /usr/lib/asterisk/modules/app_rxfax.soDec 27 12:14:27 WARNING[14679]: loader.c:355 __load_resource: No description in module /usr/lib/asterisk/modules/app_rxfax.soDec 27 12:14:27 WARNING[14679]: loader.c:362 __load_resource: No key routine in module /usr/lib/asterisk/modules/app_rxfax.soDec 27 12:14:27 WARNING[14679]: loader.c:371 __load_resource: Key routine returned NULL in module /usr/lib/asterisk/modules/app_rxfax.soDec 27 12:14:27 WARNING[14679]: loader.c:380 __load_resource: 6 errors loading module /usr/lib/asterisk/modules/app_rxfax.so, abortedDec 27 12:14:27 WARNING[14679]: loader.c:499 load_modules: Loading module app_rxfax.so failed![EMAIL PROTECTED] modules]# Any ideas? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp fax
Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1.2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] caller id
Hello all,
All my sip users are identified by their
name.lastname (mine would be dov.bigio).
But I have to associate them to extension numbers
too, so I did the following on my extensions.conf.
The problem is that when a call is logged on CDR
and also the caller ids that appear for end users is without the "."
(dot).
So if I call someone, this person would see a call
coming from "dovbigio". And he won't be able to call back to me, since
"dovbigio" is not a valid user.
Is this some kind of but, or I am doing something
wrong here?
Thank you
Dov
--
[default]
exten =
435,1,Goto(01.ramais_nomes,dov.bigio,1)
[ramais_nomes]
exten = dov.bigio,1,Macro(ramais,dov.bigio,435)
[macro-ramais]exten = s,1,SetCallerID(${CALLERID}|a)exten =
s,2,SetCDRUserField(INTERNA)exten = s,3,Dial(SIP/${ARG1},15,r)exten
= s,4,VoiceMail(u${ARG2})exten = s,5,Hangupexten =
s,104,VoiceMail(b${ARG2})Exten =
s,105,Hangup
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[Asterisk-Users] agentcallbacklogin
Hi, On of my agents made a mistake while logging in to the Queue system, and entered another agent's extension. Asterisk allowed that, and the first agent was then able to receive two calls from the queue, on that was actually for him, and the other one that was on behalf of the agent that made the mistake. Shouldn't Asterisk block the second agent in case he tries to login using an extension that is already in use by other agent? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] recording queue calls
Hi, When I set "monitor-format=wav49" on file queues.conf for a queue, Asterisk records callsat /var/spool/asterisk/monitor. But the file names it users are the call-ids of the calls. Is there a way to change that, and use information such as date, time, agent and queue to "build" the filename? It would make the localization of such files much more easy. Other useful that I miss is the capability to to allow the files to be stored in different directories, such as /var/spool/asterisk/monitor/queue1, /var/spool/asterisk/monitor/queue2, and so on, based on the queuename. Is this possible by any means? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] show queue
What is the meaning of a SL greater than 100%? lv09*CLI show queue cobrancacobranca has 0 calls (max unlimited) in 'leastrecent' strategy (6s holdtime), W:0, C:69, A:2, SL:102.9% within 45s Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] music on hold problem
I had MOH working with 1.0.9, but now it keeps showing the following log message Dec 20 11:45:05 WARNING[30548]: interface.c:215 decodeMP3: Junk at the beginning of frame 54414700 And no moh is being played! - Original Message - From: Fredrik Emil Jensen To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, December 20, 2005 6:03 AM Subject: RE: [Asterisk-Users] music on hold problem I got the same problem, look in the thread ztdummy / timer problem with kernel 2.6.14. But when I compile a new kernel back to 2.4.31 I managed to play the music for some more secs, and the shoutcast music is working fine. If you do a zttest which results do you get? And what kernel version are you running? Regards, Fredrik Jensen From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MattSent: 20. desember 2005 08:48To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] music on hold problem hi guys: my asterisk 1.2.1 just won't play music on hold, it will play a tiny bit of music at the beginning, then go silent, doing: set debug 1, found error msg: monmp3thread: Only wrote -1 of 1600 bytes to pipe: (11)Resource temporarily unavailable is this a bug? anyone know what's wrong. matt ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] meet me room status
Hi, Is there a CLI or manager command that allow me to know whether a meet me room is locked or unlocked? lv09*CLI meetme list 3User #: 02 herbertarauj Herbert Araujo Channel: SIP/herbert.araujo-0929 (unmonitored)1 users in that conference. Thank youDov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Junk at the beginning of frame
Hello users, What is the meaning of this message? Dec 19 09:19:28 WARNING[15112]: interface.c:215 decodeMP3: Junk at the beginning of frame 54414700 Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] again - show queue info
Hello list, I am comparing results of "show queue myqueue" with data from /var/log/asterisk/queue_log and have some doubts about it. When I run "show queue myqueue", I get a value of 3 for the number of abandoned. When I check the queue_log file, I have 3 calls with status "EXITWITHTIMEOUT". This way I have realized that A means "unAnswered" and not actually "Abandoned". ( I GUESS EXITWITHKEY calls would also increment the value of A). My queue has a relatively short time out (45 secs) and then the caller redirected to a voicemail. I have developed a real time monitoring application that is not handling events yet, it is simply sending manager commands and printing out the results, and my call center managers are not satisfied with the information I am currently displaying, so, handling event is certainly the best way to accomplish my goals. Does any body else have comments about this statistics, and ways of showing good real time information for call center managers? Thank you Dov -- exten = cobrancainfo,1,Answerexten = cobrancainfo,2,Queue(infocadastrais|tT|||45)exten = cobrancainfo,3,Wait(3)exten = cobrancainfo,4,VoiceMail(u501)exten = cobrancainfo,5,Hangup -- lv09*CLI show queue infocadastraisinfocadastra has 0 calls (max unlimited) in 'leastrecent' strategy (0s holdtime), W:0, C:0, A:3, SL:0.0% within 45s Members: Agent/5132 (Unavailable) has taken no calls yet Agent/4952 (Not in use) has taken no calls yet Agent/2732 (Unavailable) has taken no calls yet Agent/2462 (Unavailable) has taken no calls yet No Callers 1134644282|1134644268.462750|infocadastrais|NONE|ENTERQUEUE||pabx1134644328|1134644268.462750|infocadastrais|NONE|EXITWITHTIMEOUT|11134644358|1134644344.463504|infocadastrais|NONE|ENTERQUEUE||pabx1134644410|1134644344.463504|infocadastrais|NONE|EXITWITHTIMEOUT|11134644470|1134644456.464234|infocadastrais|NONE|ENTERQUEUE||pabx1134644516|1134644456.464234|infocadastrais|NONE|EXITWITHTIMEOUT|1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EXITWITHQUEUE on queue_log
Is there way to disable the possibility of a user leave a queue by pressing a queue? I have several occurences of EXITWITHKEY in my queue_log that shouldn't occur... 1134642743|1134642524.462637|cobranca|Agent/5230|TRANSFER|350|default1134642743|1134642524.462637|cobranca|NONE|EXITWITHKEY||1 1134646015|1134645980.464421|cobranca|NONE|ENTERQUEUE||343 1134646035|1134645980.464421|cobranca|Agent/5100|CONNECT|20 1134646171|1134645980.464421|cobranca|Agent/5100|COMPLETEAGENT|20|1361134646171|1134645980.464421|cobranca|NONE|EXITWITHKEY||1 Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue_log Vs show queue abandon calls discrepancy
Hi, Yesterday was the first day my call center operated under Asterisk 1.2.1. At the end of the day, I ran a "show queue queuename" and saw that the value of abandoned calls was 45. This morning, after updating my database with data from queue_log file, I saw, through Asterisk Guru Queue Stat, that I had only 33 abandoned calls. I tend to believe that queue_log and AsteriskGuru are more correct, because on some of the several times I tested the queue and abandoned it before being answered, I realized that the "show queue queuename" A: counter was incremented by 2. Has anyone realized such a problem? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queues music on hold
Hello list,
I have the following problem.
The behavior of music on hold is not constant on my
queues... Sometimes it plays well, sometimes it becomes mute in the middle of
the wait and sometimes it doesn't even start.
mpg123 is installed on my server.
Is there something I am missing???
Thank you!Dov
---
queues.conf
[infocadastrais]leavewhenempty=yesjoinempty=nomusiconhold=fila
strategy=leastrecent
timeout=14
eventwhencalled=yesmaxlen=0retry=0wrapuptime=5servicelevel=45monitor-format=wav49monitor-join=yesannounce-holdtime=no
member = Agent/5132
agents.conf
[agents]
autologoff=
ackcall=no
wrapuptime=5000
musiconhold = fila
recordagentcalls=no
updatecdr=yes
group =1
agent = 5132,1234
extensions.conf
exten = cobrancainfo,1,NoOp(Ligacao para Fila
de Info Cadastrais)exten = cobrancainfo,2,SetVar(prioridade=0)exten
= cobrancainfo,3,SetCIDName(CobrancaInfoCadastrais ${CALLERIDNAME})exten
= cobrancainfo,4,SetVar(QUEUE_PRIO=${prioridade})exten =
cobrancainfo,5,Answerexten =
cobrancainfo,6,Queue(infocadastrais|tT|||45)exten =
cobrancainfo,7,Wait(3)exten = cobrancainfo,8,VoiceMail(u501)exten
= cobrancainfo,9,Hangup
musiconhold.conf
[classes]fila =
mp3:/var/lib/asterisk/mohmp3/defaultfila
[moh_files]fila
=/var/lib/asterisk/mohmp3/defaultfila,r
And on /var/lib/asterisk/mohmp3/defaultfila I have
3 valid MP3 files.
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[Asterisk-Users] format_mp3 uninstalling mpg123
Hi all, In order to fix my problem with music on hold I would like to test format_mp3, that comes with asterisk-addons package. For that, the wiki says "Be sure to remove mpg123 from your system (this may attribute to 'Request to schedule in the past!?!?!' messages). Now you are set! " How do I uninstall mpg123? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] format_mp3 uninstalling mpg123
On Tue, Dec 13, 2005 at 05:12:44PM -0200, Dov Bigio wrote: For that, the wiki says Be sure to remove mpg123 from your system (this may attribute to 'Request to schedule in the past!?!?!' messages). Now you are set! How do I uninstall mpg123? How did you install mpg123? If you installed it with the package management system, then use the package management system on your OS to remove it. If you installed it manually, you'll need to remove it manually. Actually I did it manually (tar -xvzf)... but I am not sure which files I have to delete manually.. is there an explanation somehere? I couldn't find it on Google... To actually allow format_mp3 to work you also need to change musiconhold.conf from mode=quietmp3 to mode=files. This is new for me... I didn't find any information on this mode parameter... Should it be put under [classes] or [moh_files] in musiconhold.conf??? Hope that helps Thank you very much! Dov --- Gil Kloepfer [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] persistentagents, persistentmembers
Is there a way to persist agent statuses after a restart? Support I have to restart Asterisk for some reason, is it possible that all logged in (AgentCallBackLogin) would remain logged in? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] show queue in BE
Hi,
I am using Asterisk Business Edition A.1.6 (but I
guess it is the same logic for 1.2)
I am running the show queue
command for a queue that had a 36 calls and the C: parameter is growing up very
fastly, no reflecting the real calls to this queue.
lv09*CLI show queue cobranca
cobranca has 0 calls (max unlimited) in 'leastrecent' strategy (32s
holdtime), W:0, C:1006994, A:16, SL:0.0% within 45s
Here is my queues.conf
[cobranca]musiconhold=filajoinempty=yesstrategy=leastrecenteventwhencalled=yestimeout=14maxlen=0retry=0servicelevel=45wrapuptime=5announce-holdtime=nomember
= Agent/5120member = Agent/5130member = Agent/5410member
= Agent/5100member = Agent/2110member = Agent/5420
My agents.conf is
[agents]autologoff=150ackcall=nowrapuptime=5000musiconhold
=
filaupdatecdr=yesrecordagentcalls=norecordformat=wav49savecallsin=/home/asterisk/spool/monitorgroup
= 1 ; fila cobrancaagent = 5120,1234,Alessandra Barrosagent =
5130,1234,Ana Paula Furuyaagent = 5410,1234,Ana Silvaagent =
5100,1234,Bruno Tolentino Alvesagent = 2110,1234,Debora
Goncalvesagent = 5420,1234,Fabiana Montera
My extensions.conf for entering the queue:
exten = cobranca,1,NoOp(Ligacao para Fila de Cobranca)exten =
cobranca,2,SetVar(prioridade=0)exten = cobranca,3,SetCIDName(Cobranca
${CALLERIDNAME})exten =
cobranca,4,SetVar(QUEUE_PRIO=${prioridade})exten =
cobranca,5,Answerexten = cobranca,6,Queue(cobranca|tT|||50)exten
= cobranca,7,Hangup
This is very important since is it is preventing my Call Center Monitoring
application to work (it worked well while running 1.0.9 open source).
Does the C: value mean a different thing? Or is there any configuration that
I am missing somewhere?
Thank you very much
Dov
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Re: [Asterisk-Users] Error on using queue.
If you are using 1.2, it might be the joinempty and
leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 11:27
AM
Subject: [Asterisk-Users] Error on using
queue.
I am trying to use * as ACD server for our sip
proxy.
I first dial 55 to login 98 as
ACD agent it worked fine and then when I dialed 98, I got these messages from * CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new
stack -- Executing Wait("SIP/98-f718", "2") in
new stack -- Executing Queue("SIP/98-f718",
"queue1") in new stackNov 30 16:54:12 WARNING[7579]: app_queue.c:3078
queue_exec: Unable to join queue 'queue1' -- Executing
Hangup("SIP/98-f718", "") in new stack == Spawn extension
(default, 99, 5) exited non-zero on
'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen
= 0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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[Asterisk-Users] meet me message
Since upgrade to BE A.1-6I get the following messages on my console... -- x=0, open writing: /var/spool/asterisk/meetme/meetme-username-2-3 format: sln, 0x9e454b8 And several .sln files are saved on /var/spool/asterisk/meetme/ What do this mean? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error on using queue.
How is your agents.conf ? How is your login in
extensions.conf?
- Original Message -
From:
gc
To: Dov Bigio ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 2:53
PM
Subject: Re: [Asterisk-Users] Error on
using queue.
Thanks. I made change to joinempty=yes. And now I
can hear the music on hold. But it would not ring the agent even if I login
agent in. When I run show queue command under CLI, I got these
messages:
queue1 has 1
calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:0, A:2,
SL:0.0% within 0s Members:
Agent/555997 (Unavailable) has taken no calls
yet Agent/555998 (Unavailable) has taken
no calls yet
It seems that something wrong with my config
file, it did not login any agent.
- Original Message -
From:
Dov Bigio
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, December 01, 2005 8:33
AM
Subject: Re: [Asterisk-Users] Error on
using queue.
If you are using 1.2, it might be the joinempty
and leavewhenempty parameters.
Their default are different than the 1.0.x
releases
- Original Message -
From:
gc
To: Asterisk Users Mailing List
- Non-Commercial Discussion
Sent: Thursday, December 01, 2005
11:27 AM
Subject: [Asterisk-Users] Error on
using queue.
I am trying to use * as ACD server for our
sip proxy.
I first dial 55 to login 98
as ACD agent it worked fine and then when I dialed 98,
I got these messages from *
CLI:
-- Executing
Answer("SIP/98-f718", "") in new stack --
Executing Ringing("SIP/98-f718", "") in new
stack -- Executing Wait("SIP/98-f718", "2")
in new stack -- Executing
Queue("SIP/98-f718", "queue1") in new stackNov 30 16:54:12
WARNING[7579]: app_queue.c:3078 queue_exec: Unable to join queue
'queue1' -- Executing Hangup("SIP/98-f718",
"") in new stack == Spawn extension (default, 99, 5)
exited non-zero on 'SIP/5025155598-f718'
Can anybody tell me what cause this
problem?
The followings are my configuration
files:
extensions.conf:
[default]
;For incoming call to ring into the
queue.exten= 99,1,Answerexten=
99,2,Ringingexten= 99,3,Wait(2)exten=
99,4,Queue(queue1)exten= 99,5,Hangup
;Agent loginexten =
55,1,AgentCallBackLogin(|[EMAIL PROTECTED]);Agent
logoutexten = 55,1,AgentCallBackLogin(|1)
exten =
97,1,Dial(SIP/97)exten =
98,1,Dial(SIP/98)
agents.conf:
[Agent1]agent =
97,,Gary1agent = 98,,Gary2
queues.conf:
[queue1]musiconhold =
defaultstrategy=ringalltimeout=15retry=5wrapuptime=0maxlen
= 0announce-frequency = 0announce-holdtime = nomember =
Agent1/555997member = Agent1/555998
sip.conf:
port=5060bindaddr=192.168.111.11context=defaultallow=ulaw
[97]type=friendusername=97insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
[98]type=friendusername=98insecure=verycanreinvite=nocontext=defaulthost=192.168.111.2
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[Asterisk-Users] Fw: asterisk shutting down...
Hi, Got the following messages log tonight... and Asterisk was down until I manually restarted it... Any ideas? Thank you Dov Oct 19 03:40:18 WARNING[28005]: Avoided deadlock for 'SIP/raphael.pavanelli-f40b', 10 retries!Oct 19 03:40:28 NOTICE[28005]: Still have a call...Oct 19 03:40:50 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:40:50 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:44:21 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:44:59 WARNING[28005]: Avoided initial deadlock for 'SIP/marcelo.araujo-0241', 10 retries!Oct 19 03:45:31 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:46:41 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:46:51 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:47:46 WARNING[28005]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)Oct 19 03:47:49 WARNING[28005]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)Oct 19 03:47:57 WARNING[28005]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)Oct 19 03:48:00 WARNING[28005]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request)Oct 19 03:48:01 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:49:28 NOTICE[28005]: Still have a call...Oct 19 03:50:12 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:50:12 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:50:32 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:53:53 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:53:56 WARNING[28005]: Avoided deadlock for 'SIP/raphael.pavanelli-f40b', 10 retries!Oct 19 03:54:54 WARNING[28005]: Avoided deadlock for 'SIP/alexandre.catao-d9b5', 10 retries!Oct 19 03:55:03 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:55:03 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:56:13 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:56:13 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] soxmix generating mute files
Hello All, I am trying to use soxmix to merge two wav files generated by monitoring calls from a queue, since it generated two files (in out). When I run soxmix file1.wav file2.wav mixedfile.wav, although file1.wav and file2.wav are good, mixedfile.wav is file with the same size as file2.wav, but totally mute. Any clues? Thank you!Dov ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] xpro codecs and asterisk
Hi all, I am trying to make a call from an X-Pro with only the G.729 codec enabled to another with both G.711 and g.729. The Asterisk version is 1.0.3 and canreeinvite is set to Yes. What happens is that I got an 403 - Forbidden response and setting verbose 10 in Asterisk I can see the message: May 2 15:47:36 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/victor-a02d(4) to SIP/dediana-1fd9(256) -- SIP/victor-a02d is ringing -- SIP/victor-a02d answered SIP/dediana-1fd9May 2 15:47:38 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/dediana-1fd9(256) to SIP/victor-a02d(4)May 2 15:47:38 WARNING[6690]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/dediana-1fd9 compatible with SIP/victor-a02d If I do the same when both softphones have only G.711 set, everything works fine. It seems that Asterisk tries to use the first codec in SDP and ignores others. Does it make sense? Thanks in advance. Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] codecs, asterisk, xpro
Hi all, I am trying to make a call from an X-Pro with only G.729 codec enabled to another with both G.711 and g.729. The Asterisk version is 1.0.3 and canreeinvite is set to Yes. What happens is that I get an 403 - Forbidden response and setting verbose 10 in Asterisk I can see the message: May 2 15:47:36 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/victor-a02d(4) to SIP/dediana-1fd9(256) -- SIP/victor-a02d is ringing -- SIP/victor-a02d answered SIP/dediana-1fd9May 2 15:47:38 WARNING[6690]: channel.c:2115 ast_channel_make_compatible: No path to translate from SIP/dediana-1fd9(256) to SIP/victor-a02d(4)May 2 15:47:38 WARNING[6690]: app_dial.c:1006 dial_exec: Had to drop call because I couldn't make SIP/dediana-1fd9 compatible with SIP/victor-a02d If I do the same when both softphones have only G.711 set, everything works fine. It seems that Asterisk tries to use the first codec in SDP and ignores others. Does this make sense? Thanks in advance. Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] queue - transfer calls
Hello, I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation. We have a call center with 4 agents, which should receive calls from their queue. But we also have a "call center management" team which should be able to talk to end customers in case the first level call center is not able to solve the problem. There are two issues there: 1. The agent cannot use the soft-phone TRANSFER button.. she has to press the pound key to transfer. This is not a 'terrible' issue, since it is just a matter of educating agents. 2. Attended transfer: If the agent transfers the call to someone in the management team, the call is immediately transferred, and the agent is not able to talk to the manager before. Is there a way to allow an agent to talk to the management befora actually transferring, so that he can explain the issue in advance Thank you very much Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re:queue - transfer calls
Thanks Ariel. Your 2nd suggestions seems a good bypass for this problem... it might be helpful here, thanks! About the 1st one (using paid X-Ten software), I am using paid X-Pro, which does have a transfer button... but ifIuse this button instead of pound, the calls simply hangs up.. But I think that unfortunately, this is the expected behaviour! ThanksDov From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Dov BigioSent: Monday, April 18, 2005 9:16 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Hello,I am setting up an ACD using *, but found a an issue that I am not beingable to resolve, and this might impact our * implementation.We have a call center with 4 agents, which should receive calls from theirqueue. But we also have a "call center management" team which should be able to talk to end customers in case the first level call center is not able tosolve the problem.There are two issues there:1. The agent cannot use the soft-phone TRANSFER button.. she has topress the pound key to transfer. This is not a 'terrible' issue, since it isjust a matter of educating agents.This one can be fixed if you want by going with the paid xten pro software.It has a transfer button.2. Attended transfer: If the agent transfers the call to someone in themanagement team, the call is immediately transferred, and the agent is notable to talk to the manager before. Is there a way to allow an agent to talkto the management befora actually transferring, so that he can explain theissue in advanceIn stead of transferring to the next level support have your agents park thecall to lets say 700 it should give you something like 701 then call thenext agent tell them what the problem is and to pickup exten 701.Thank you very muchDov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: queue - transfer calls
Hi Ariel, Thinking a little bit more about your idea of parking calls for 'simulating' a consultive transfer, I realized the following problem: If an agent is making an outgoing call (or even receiving a call that is not coming from the queue), he is not considered busy to the queue manager.] That means that once the agent parks a users call, if calls to his manager to tell him there is a parked call waiting to be answered, he immediately becomes available to the queue, and might receive calls even while he is talking to the manager. Is there a way to define that an agent is busy if he is on any call, not just calls coming from the queue? Thank you Dov Message: 9Date: Mon, 18 Apr 2005 10:18:31 -0400From: "Ariel Batista" [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] queue - transfer callsTo: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset="us-ascii"Hello,I am setting up an ACD using *, but found a an issue that I am not being able to resolve, and this might impact our * implementation.We have a call center with 4 agents, which should receive calls from theirqueue. But we also have a "call center management" team which should be able to talk to end customers in case the first level call center is not able tosolve the problem.There are two issues there:1. The agent cannot use the soft-phone TRANSFER button.. she has topress the pound key to transfer. This is not a 'terrible' issue, since it isjust a matter of educating agents.This one can be fixed if you want by going with the paid xten pro software.It has a transfer button.2. Attended transfer: If the agent transfers the call to someone in themanagement team, the call is immediately transferred, and the agent is notable to talk to the manager before. Is there a way to allow an agent to talkto the management befora actually transferring, so that he can explain theissue in advanceIn stead of transferring to the next level support have your agents park thecall to lets say 700 it should give you something like 701 then call thenext agent tell them what the problem is and to pickup exten 701.Thank you very muchDov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zombie channels missed transfer
Hello, I have an Asterisk server and Audio Codes installed in my network, talking to my legacy PBX. If I have a call between 2 soft-phones, and one of them wants to do a supervised transfer of the other to a legacy PBX extension, what happens is that: 1. Person #1 opens a new channel with the PBX extension; person #2 is on hold (listening to Music On Hold) 2. Person #1 talks to PBX user 3. Person #1 hits the TRANSFER button (on X-Pro) and the channel that was open with #2. 4. Person #2 is mute, and PBX keeps on listening to Music On Hold??? 5. If PBX hangs up, #2 is also hung up.??? If I see the results of "show channels" between steps 4 and 5, I get a ZOMBIE channel. Does anyone knows more about this behavior? Step 4 is very strange and prevents me from transfering VoIP calls to a PBX user... Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, ACD, Queues and Call Transfer Issue
Hello, I have implemented a test ACD with Asterisk 1.0.7, in which I have 2 agents and one user making calls and using AgentCallbackLogin. Besides that, I have other users on the PBX, but not necessarily members of any queue. Agents and PBX users are using X-Pro as a soft-phone. I am having problems in the case where one agent answers the call and for any reason needs to transfer this call to the other agent, or to somebody else in the PBX system. When the agent clicks the transfer button on the soft-phone, the call is hang up and we loose the client. If the agent dials '#' then the transfer works fine. The problem is that in this case the person to whom the call is being transferred must have a numerical extension (which we didn't want to use internally). is there a solution for this? Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk sounds
Hello all, I am looking for a list of all available sound files for asterisk and a transcription of their content, so that I can have someone translate them into portuguese. Does anybody have a list of these files? Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk sounds
Thanks a lot!!! De: "Josiah Bryan" [EMAIL PROTECTED] Para: "David John Walsh" [EMAIL PROTECTED],"Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Cópia: [EMAIL PROTECTED] Data: Tue, 5 Apr 2005 14:52:58 -0500 Assunto: Re: [Asterisk-Users] asterisk sounds On Tuesday 05 April 2005 3:28 pm, David John Walsh wrote: Dov, If anyone responds to your request privately, I'd apreciate it if you were to forward it to me, as I need to translate them into several european launguages. Guys - As others more enlightened than myself pointed out - Look at /usr/src/asterisk/sounds.txt, where /usr/src is the location of your asterisk CVS tree. sounds.txt has both the file name and the transcript of the audio. -josiah -- Josiah Bryan IT Coordinator Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] configuring md5 authentication
Hello, How does md5 authentication works? I have created a user on my sip.conf like this: [dov]type=friendhost=dynamicusername=dovauth=md5; echo -n "dov:myhost.com.br:dov" | md5summd5secret=a72d3b44ea28fc6515d922b21970b83c ;secret=dov Where myhost is the real that I normally use on my SIP phone when I don't use md5 authentication. The echo line is the command I used to convert my user:realm:pwd into md5. In my X-Ten phone I just enter my username "dov" and password "dov" as plain text. It doesn't log in as I thought it should... Is there any extra setting that I have to define? Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sharing asterisk among several companies
Hello, I am trying to configure Asterisk to be used by two (or more) different remote companies, sharing the same instance of Asterisk on my host. By setting specific entry contexts for each sip user, I can repeat extensions among companies. My question is: is it possible to have repeated users on sip.conf being identified by their different passwords? I tried to do that but got an authentication failure. Is there a way to do this? Or I should always have different usernames? Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] adding extension ChanSpy
Hi ALL, I have downloaded app_chanspy.c and chanspy_sounds.tgz. But I haven't found any instructions on how to compile and where to untar these files... I tried to put the .c file on asterisk-src/apps and remake asterisk, but it seems it was not enough... Thank you!Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] music on hold issue
Hello,I am configuring an ACD and have created queues, extensions and agents correctly (at least basic functionality is working).I added an mp3 file to the /var/lib/asterisk/mohmp3/ directory and configured it on agents.conf When I user calls and is on the queue, he starts listening to the mp3 file perfectly. But then the music stops, and only continues while the user speaks on the microphone. That is, the music goes on as the user speaks and not continuously.Does anybody have any clue about this?I read something about configuring timing, using a zaptel or usb (i have an usb in my dev't environment), but I don't know if this is related to my problem and couldn't find details on how to do itThank youDov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] slim server for moh
Hello, I have installed SlimServer for Windows on my desktop and Asterisk on a Red Hat Linux machine. I am able to play mp3's for music on hold when mp3s are on the Linux server, and to play streaming mp3's with Windows Media Player and Winampon Windowsusing the slim server. I also have mpg123 on my Linux, apparently installed correctly, since it works for local moh. I put the following line on my musiconhold.conf default=custom:/var/lib/asterisk/mohmp3-dummy,/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 http://10.0.0.90:9000/stream.mp3 where 10.0.0.90:9000 is my slim server. On /var/lib/asterisk/mohmp3-dummy I have an empty mp3 file with 0bytes. This doesn't work at all (for example using the following extension) exten = 64,1,Answerexten = 64,2,MusicOnHold(default)exten = 64,3,Hangup I have converted my mp3 files so that they have the following characteristics )but I don't really think this matters since when the music is on the Linux machine, without slim server, it works: MPEG 2.5 layer 316kbit, 1385 frames8000Hz Mono Any help would be really appreciated!!! Thank you Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
