Re: [asterisk-users] asterisk crash and core dump
On Tuesday 18 September 2007 15:15:38 Vieri wrote: My Asterisk installation crashes frequently. Since it's a random event I am not able to reproduce it so I can't say what is making it crash. Here's a snippet of /var/log/asterisk/full just when it crashes: Sep 18 13:42:51 DEBUG[378] chan_zap.c: disabled echo cancellation on channel 31 Sep 18 13:42:51 VERBOSE[378] logger.c: -- Hungup 'Zap/31-1' Sep 18 13:42:51 DEBUG[32650] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:52 DEBUG[32677] manager.c: Manager received command 'Command' Sep 18 13:42:54 DEBUG[32650] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Sep 18 13:42:54 VERBOSE[419] logger.c: -- SIP/4002-082aef20 is ringing ---MESSAGE FROM SAFE_ASTERISK--- Automatically restarting Asterisk. Sep 18 13:43:03 VERBOSE[551] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf': Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/dnsmgr.conf' : Found Sep 18 13:43:03 VERBOSE[551] logger.c: Asterisk Dynamic Loader loading preload modules: Sep 18 13:43:03 VERBOSE[551] logger.c: == Parsing '/etc/asterisk/modules.conf' safe_asterisk, ie FreePBX, notifies me that Asterisk exited on signal 11. I have core dumps in /tmp. What can I do to isolate the cause of these segmentation faults? You should compile asterisk with debbuging enabled (and optimization disabled), and then take backtraces from core dumps. Please see http://www.voip-info.org/wiki-Asterisk+debugging Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Center SoftPhone with Auto Answer
On 9/18/07, Joao Pereira [EMAIL PROTECTED] wrote: I don't think so, because in paging/intercom, the phones must support Auto Answer. The link you sent says: SIP phones for the most part don't support any of these phone based paging functions. If a SIP phone offers an Auto Answer function, you can approximate limited paging intercom functionality. I'm using X-Lite, and in X-Lite I can't force the users to answer the call. The users can put Auto Answer = Off. Also, the response from Counterpath was weird, as they said they're engineering team cannot remove the Auto Answer option: To have the auto-answer permanently on in the context that you wish to have is a feature that our engineering team cannot hard code into the phone. It can be turned on and off in the menu Actually i believe you can do it yourself. X-Lite is windows, right? There are a bunch of programs, allowing to edit internal resources of executable files. So, just grab a resource editor (i prefer XN Resource Editor), open .exe file, edit the menu - disable (and hide) items you want to forbid changing for users, and give them the executable. I'm not certain that X-Lite's executable is not packed/crypted, but editing SJPhone was very successful some time ago. Of course, there's always an option for user - to take another softphone, but whatever softphone you choose - they will have the same chance. Regards, Atis ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCalbackLogin not loging in race condition ?
On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote: Previous mail did not go through. Following up... Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit: Hi, I'm seeing a problem using AgentCallbackLogin (Asterisk 1.2.16) where a call in queue while an agent is logging in results in the agent getting the call without properly being logged in. This seems to be a race, although I've not (yet) pinpointed the code at fault. And I'm not able to reproduce it 100% of the time. The perceived anomaly is that teh agent is logged of w/o request, but it seems it never got logged in. This only happens when logging in with calls already in queue. Any hints ? I'm going to make AgentCallbackLogin set the initial state to wrap to see if that patches the problem by the time being. Any hints are welcome. I did not find a way to go wrap on login, cause the logic is distributed between Queue and Agent, but I did find this comment at chan_agent.c: /* Ensure we can't be gotten until we're done */ gettimeofday(p-lastdisc, NULL); p-lastdisc.tv_sec++; It seems that the time it takes for the login ok message is more than one second, and is creating the trouble window. I changed that to allow 10 seconds of unavailability and the problem seems to be gone. -Carlos Shouldn't wrapuptime be used in this case? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCalbackLogin not loging in race condition ?
On Wednesday 19 September 2007 12:11:19 Carlos G Mendioroz wrote: Atis Lezdins @ 19/09/2007 06:05 -0300 dixit: On Wednesday 19 September 2007 11:43:39 Carlos G Mendioroz wrote: Previous mail did not go through. Following up... Carlos G Mendioroz @ 16/09/2007 13:27 -0300 dixit: Hi, I'm seeing a problem using AgentCallbackLogin (Asterisk 1.2.16) where a call in queue while an agent is logging in results in the agent getting the call without properly being logged in. This seems to be a race, although I've not (yet) pinpointed the code at fault. And I'm not able to reproduce it 100% of the time. The perceived anomaly is that teh agent is logged of w/o request, but it seems it never got logged in. This only happens when logging in with calls already in queue. Any hints ? I'm going to make AgentCallbackLogin set the initial state to wrap to see if that patches the problem by the time being. Any hints are welcome. I did not find a way to go wrap on login, cause the logic is distributed between Queue and Agent, but I did find this comment at chan_agent.c: /* Ensure we can't be gotten until we're done */ gettimeofday(p-lastdisc, NULL); p-lastdisc.tv_sec++; It seems that the time it takes for the login ok message is more than one second, and is creating the trouble window. I changed that to allow 10 seconds of unavailability and the problem seems to be gone. -Carlos Shouldn't wrapuptime be used in this case? Regards, Atis This is happening at login time. wrapuptime gets used after the agent handles a call. I do have wrapuptime set BTW. wrapuptime uses lastcall as reference, which is a Queue var. Login is an Agent process. I still do not understand the whole thing though... Ok, my mistake. AgentCallbackLoging really doesn't add to queue. So, your agent is already in queue, when you do AgentCallbackLogin? Or how otherwise would you get call from queue? I'm not sure for static members, i'm using QueueAdd after AgentCallbacklogin. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue serializes call delivery ?
On Wednesday 19 September 2007 13:03:30 Carlos G Mendioroz wrote: This might be obvious, and well known, but... If I have 5 ready members and 5 calls in queue at once, Queue seems to deliver them one by one, blocking while waiting for each member to answer in turn. Is there anyway to speed this up (other than setting auto answer ?) I.e., I would like to have paralel calls to 5 members if I have 5 calls in queue... This is available starting from 1.4, see UPGRADE.txt: * The old/current behavior of app_queue has a serial type behavior in that the queue will make all waiting callers wait in the queue even if there is more than one available member ready to take calls until the head caller is connected with the member they were trying to get to. The next waiting caller in line then becomes the head caller, and they are then connected with the next available member and all available members and waiting callers waits while this happens. This cycle continues until there are no more available members or waiting callers, whichever comes first. The new behavior, enabled by setting autofill=yes in queues.conf either at the [general] level to default for all queues or to set on a per-queue level, makes sure that when the waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers, whichever comes first. This is probably more along the lines of how one would expect a queue should work and in most cases, you will want to enable this new behavior. If you do not specify or comment out this option, it will default to no to keep backward compatability with the old behavior. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to cancel the password check in VoicemailMain()
On Wednesday 19 September 2007 19:03:18 Mark Michelson wrote: ur VoiceMailMain call. Change it to this and see if it helps: exten = 99,n,VoiceMailMain([EMAIL PROTECTED]) In other words, put the 's' at the beginning of the argument as opposed to a separate option. I think, it is deprecated in 1.4, and should work at the end. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk crash and core dump: format_mp3.so
On Thursday 20 September 2007 11:34:44 Vieri wrote: My Asterisk server process irregularly segfaults, ie. it usually works fine (is stable) when there's low traffic but repeatedly crashes during morning hours when there are more calls. I gdb'ed the core dump files and found that the culprit may be format_mp3. So I disabled MOH today and will see if that's the cause. I know that mp3 files are known to cause * crashes but what I don't understand is why it doesn't *always* crash (ie. why doesn't it crash even when there's low traffic? I mean, if the offending code is in the mp3 format then it should *always* crash, right?). We also experienced this problem on 1.2, but i'm not sure that this is registered in bug database. You should check bugs.digium.com and if it's still valid for 1.4, you should post your backtraces there. As solution - we refused from using format_mp3 at all - actually it has almost no benefits. If your MOH is in MP3s - you will get them decoded (and translated to necessary codec) on-the-fly for every call, so more performance. You can convert all your MOH to native channell formats of asterisk, and put all those files (one for each format/MOH combination) in MOH directory - asterisk will pick up one with less translation. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and MS Exchange 2007
On Friday 21 September 2007 15:55:14 Olivier wrote: Hi, Here you can find a list of MS Exchange 2007 compliant systems: http://www.microsoft.com/technet/prodtechnol/exchange/telephony-advisor.msp x You cannot see mention of Asterisk ;-)) Has anyone tried to integrate Asterisk and Exchange 2007 ? A prospective customer is using MS Exchange 2007 and is asking for such integration. Passion aside, what do you think of such integration ? Would you expect a better user experience from an Asterisk-Exchange bundle than a pure Asterisk ? I like the sentence: Since PBXs are so diverse and proprietary, PBX planning is probably the most difficult pre-deployment task. :D Well, i believe everything that needs to be done is creating configuration notes, and sharing them, as here: http://www.microsoft.com/technet/prodtechnol/exchange/pbx-partners.mspx I just browsed some of direct SIP configuration, they don't seem complex. So, if you got the exchange, just play with it, and send us and microsoft the notes (in oo.org doc ;) Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Authenticate() application and CDR
On Friday 21 September 2007 19:07:43 Ricardo Carvalho wrote: Works great, although there is a problem with the CDR: Asterisk accounts only the call that is answered by Asterisk which asks for the pin. Case the call hasn't been answered by the called party, or case it even hasn't been dialed because the caller failed to insert the pin, or even if it has been answered, Asterisk writes in CDR table that it has been ANSWERED and billed from the time Asterisk picked up to ask the pin. I'm I skipping something in my syntax, or is this some kind of BUG? (I'm using Asterisk version 1.2.17) Nop, your dialplan is correct, and this is not a bug. Answer() in first line marks incoming call answered, so counter (also from your provider) is on, and you can't turn it off. Of course, Answer() is required, so that asterisk can start receiving voice, and DTMF to authenticate. If you would want to do your own billing, to count only duration of call dialed to SIP/whatever, you can do 1,Answer() 2,Authenticate() 3,Playback() 4,ResetCDR() 5,Dial() NoCDR would tell to not write CDR for that channel, but ResetCDR later would reset answer status for CDR, and start counting duration from that moment. ResetCDR(w) would make you have two CDR records, one for each part (that can be linked together by using uniqueid). Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] # to transfer calls
On Monday 24 September 2007 10:21:44 VoIP Newbie wrote: I wonder why my call was transferred when I pressed '#' in a conversation. How can I disable this kind of call transfer? Thanks. David Take a look at features.conf - probably there is blind transfer enabled on # key. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crash and debug
On Monday 24 September 2007 14:06:27 Joao Pereira wrote: Hello each 15 days my Asterisk crashes. Every time it happens I try to change something in its configuration to avoid the next crash. I already checked the logs but I don't know what to do. Can someone tell me whats the problem? These are my Asterisk logs: http://vox.fccn.pt/crash Well, this seems familiar. Notice that the first line of starting asterisk is Sep 24 09:45:56 VERBOSE[7784] logger.c: Asterisk Event Logger Started /var/log/asterisk/event_log And line before is Sep 24 09:45:51 DEBUG[14393] manager.c: Manager received command 'Command' So, you're doing some CLI command trough AMI. I guess, it's show channels ;) I've seen it a lot on 1.2 (am i correct). I get rid of that o stopped only after upgrading to 1.4.10 Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
On 9/25/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Adrian Marsh wrote: I'm interested in how people are clustering Asterisk, if that's possible, or how you might be achieving a redundant solution. I've a single Asterisk server driving the company. Its well backed-up, and I've a cloned machine that (in theory) with a DNS change could take over operations. However I'd like to achieve something more automated if possible. Maybe my post at http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Redundancy
[snip] http://lists.digium.com/pipermail/asterisk-users/2007-August/195339.html could provide you with some answers. Hi, This seems nice way of sharing settings, however it wouldn't take over calls in progress. For us, currently the greatest problem is that whenever Asterisk crashes, calls are lost, and that means - lost money. Are there any ideas? You might want to take Asterisk out of the media path then. If it crashes, calls will stay up, although your CDR's will be screwed. If screwed CDR's still means lost money... your still screwed! Nop, i can't stay out of media path, as there are essential features depending on it - hell, that's why i need asterisk - transfers, chanspy, monitoring.. Of course in case of crash - monitoring and CDR can be lost - that would be minor problem comparing to lost calls. I'm thinking about some mechanism how asterisk could communicate with second asterisk and report all state operations made with SIP. So if asterisk fails, redundancy asterisk performs IP takeover and continues. Unfortunately my SIP knowledge is nearly minimal (as are my C skills), and i don't have any ideas how to implement this. A simplest method could be something like SIP proxy, that sends calls to asterisk, but if asterisk stops responding, it plays some message and tries to send call to redundancy server - however then problem can occur with redundancy server. And this would have some major drawbacks - calls wouldn't be matched to corresponding agents in queue. Hmm, thinking a bit more about topic - maybe redundancy mechanism would have enough to keep state of channels, bridges, and corresponding dialplan location (assuming that config is identical). Too much of duplicating everything would mean that second asterisk could have the same crash. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3-way calling
On Friday 28 September 2007 09:16:14 Rilawich Ango wrote: Do u mean meetme? It is total different from my case. In meetme, everybody need to know and dial the conference room number to get into the conference room. In my case, party A,B,C may not know the conference number. A only knows B numbers and B only knows C numbers. I'm planning to do something similar, and i have created a prototype code for this. So my prototype works: 1) A dials B 2) B presses some key to launch DYNAMIC_FEATURE (features.conf) 3) the feature fires a script that joins both channels to conf room. 4) B presses some key to exit from conf, and get to specified exit context. 5) DISA() there gives a dialtone, and launches dial to C 7) B presses first key again to join both calls to the same conference. 8) B can repeat again from 4 to add more calls to conference. Now reading all this gave me idea thaht it could be better to merge 3, 4 and 5 so that if nobody is in conference, you probably want to add some more people to conference - so just don't add B there, but give DISA straight away. Also this wouldn't allow neither A or C to add somebody to the same conference, as conference's name would match B's extension - otherwise it would be hard to determine wich conference to add. Regards, Atis On 9/28/07, Pamela Weis [EMAIL PROTECTED] wrote: it is probably not what you are looking for. but simply use a conference room of asterisk for those 1 line phones. pamela ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call relation in call transfer
On Friday 28 September 2007 10:56:19 Rilawich Ango wrote:
Thanks.
Actually, I want to have some information about the call transfer just
like to queue_log in queue.
According to your message, there is no such mechanism to associate the
call in call transfer. How about any variable that I can identify the
call which is made by call transfer?
As I know there is a variable ${BLINDTRANSFER} that will fill in a
value in blind transfer. However, I can't find any variable that will
fill in a attended transfer. Anyone can advise?
Hi,
I have done this in dialplan logics. First i'm setting some global inherited
variable Set(__call_id}=${UNIQUEID}) - that is unique for channel. That
becomes call id for entire call - wherever it would gou - queues, transfers,
etc. As it's inherited it is copied to newly created channels. Then in CDR's
userfield i add ${call_id}, plus number that identifies call leg. This makes
my CDR easilly linkable and trackable.
Regards,
Atis
On 9/28/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 28 Sep 2007, Rilawich Ango wrote:
In CDR, I found that there are 3 records after doing call transfer.
However, 3 of them are individual record that is very difficult to
identify they are related to call transfer. My question is how to
identify the call with a clear flow, from CDR or by other means, is a
call transfer.
Do they have a common criterion? If they do not have a common
criterion, it is probably not logically possible to associate them.
Asterisk is a back-to-back user agent, so it builds out distinct legs for
every call with unique Call-IDs and dialogue tags. This makes it hard to
meaningfully associate call flows like this inherently, unless you do
state tracking in the software to make this possible.
This has been an ongoing topic of discussion periodically on the
Asterisk Developers' List (asterisk-dev). It seems there is considerable
interest in reworking the CDR engine to account for this type of
situation more meaningfully. You may wish to search the list archives
for greater insight into what core developers are thinking, or to join
the list and add your two cents to what you want to see from it. You're
definitely not the first person to run into this or regard it as a
serious impediment. :)
Cheers,
--
Alex Balashov
___
Sign up now for AstriCon 2007! September 25-28th.
http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue members, URI.
On Tuesday 02 October 2007 19:30:44 Thomas Kenyon wrote: Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/number@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the party. (and if need be, there is the ability in queues to run a script on connection iirc). I'm migrating to Local/number@context right now (from Agent/ channels), and it seems to me that Local channels doesn't show (busy) in show queues. This will probably require for me to do some overhead work for correctly displaying agent status in monitoring software, but i think i will be able to do it by combining core show channels with show queues. I'm not sure is it related to Agent channels that could accept only one call or SIP channel status. I would expect queue to show even Local channel as busy if there is active call trough it. I think this really can't be accomplished by dialplan logics, as dialplan is not executed upon show queues Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Resolving digit strings using pound/hash.
On Wednesday 03 October 2007 15:41:08 William F. Acker WB2FLW +1-303-722-7209 wrote: Hi all, The thing that has bugged me about Asterisk since I first started playing with it, is the fact that the pound sign/hash/octothorp doesn't resolve digit conflicts or cancel timing on a variable length string such as a tie line code or when you call numbers in a country whose length can be different between numbers in the same plan. In North America, we see this when calling places such as Germany. Thanks to Atis, this now seems to work properly in DISA fixed via bug 10754. Can we please have this effect expanded to cover all cases where Asterisk collects digits such as dialing into an IVR, zap FXS channel, and everywhere else. Hi, Can you pinpoint (with examples) where it is not that way? From my experience this is already working nearly everywhere. At least it's for Read's and incoming calls. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to get asterisk to take a dump?
On Wednesday 03 October 2007 20:48:37 Steve Edwards wrote: install -- 1.2.7.1, but it has custom code that needs to be updated before moving to a more recent release. I'm assuming that 100mb is indicative of a memory leak (probably in my code). How can I get a dump (preferably without disrupting production) so I can poke around in it (using gdb) and what's a good strategy for finding memory leaks? Thanks in advance, I think, there's no way you can get a coredump without interrupting process. However you can do killall -5 asterisk. That would send a Trace/Breakpoint signal to asterisk and it would crash immediately to core - so you can play with it in gdb. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAXy and hook flash transfer
On Wednesday 03 October 2007 22:21:24 Michael Munger wrote: In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here Any help is appreciated. Do you have t and/or T flag set in Dial() options? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agent Callback Login in 1.4
Can you describe exactly what you lose by using the dynamic queue member alternative? We tried to ensure that no functionality was lost in this transition, so if there is something that was missed please let us know what it is and we'll try to take care of it. Now, i'm finally trying to migrate, and i see a problem here. When i was using Agent channels there was status Busy indicated in show queues, whenever agent was on call from queue. I'm trying to do all the stuff with RT queue members and Local channels, but i'm missing this. I have read about GROUP usage in Local channel - so that upon call arrival Local channel can indicate that it's busy, however this is not executed upon show queues - so no status changes occur. I believe this have some connection with ast_device_state_changed, but it's only available in chan_agent, that as i understand is deprecated. Is there any other way how i would get status indication in show queues? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Injecting a sound file into a bridged call
On 08/10/2007, Girts Graudins [EMAIL PROTECTED] wrote: I'm looking for a way to play a sound file to an already established bridged call. It is meant for one party, but it's ok if both parties would hear it. Ideally, I'd like to be able to trigger this from the Management Interface with something like: I'm also in need for such functionality, the only difference is that i need for both channels to hear the message. As i have read press releases, there will be something similar available in 1.6. If you succeed, please give us a note - how it can be done. 2) I've seen whisper-type of functionality associated with meetme rooms, but I'd rather not set up a dynamic meetme room for each call I'm bridging; Well, you can create conference dynamically whenever you need to play the file. I started working on this, and have found several bugs regarding this, but they should be fixed in 1.4.12 Idea is to Redirect() trough AMI both channels to dynamical conference, and then attach call with Playback() to the same conference. For now, the Redirect() part is working fine, but due to lack of time, i haven't got further. On Monday 08 October 2007 14:13:38 Jaswinder Singh wrote: See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk to one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But i dont know how to play a recorded file in it . My collegues tried this but unsuccessfully. The basic idea is to use local channels - one is bridged to Chanspy() and second to Playback(). I'm not sure what is the problem, but theoretically also this should work. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Injecting a sound file into a bridged call
On Monday 08 October 2007 17:42:00 Dovid B wrote: Have a look here: http://www.voip-info.org/wiki-Asterisk+config+features.conf .Specifically at applicationfaturemap. You're right, that should work. I was just too much concentrated on my problem - playing sound for both channels without interrupting communications. However this will stop bridge of channels - so only one party will hear prompt, but second - silence. Regards, Atis - Original Message - From: Atis Lezdins [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, October 08, 2007 2:10 PM Subject: Re: [asterisk-users] Injecting a sound file into a bridged call On 08/10/2007, Girts Graudins [EMAIL PROTECTED] wrote: I'm looking for a way to play a sound file to an already established bridged call. It is meant for one party, but it's ok if both parties would hear it. Ideally, I'd like to be able to trigger this from the Management Interface with something like: I'm also in need for such functionality, the only difference is that i need for both channels to hear the message. As i have read press releases, there will be something similar available in 1.6. If you succeed, please give us a note - how it can be done. 2) I've seen whisper-type of functionality associated with meetme rooms, but I'd rather not set up a dynamic meetme room for each call I'm bridging; Well, you can create conference dynamically whenever you need to play the file. I started working on this, and have found several bugs regarding this, but they should be fixed in 1.4.12 Idea is to Redirect() trough AMI both channels to dynamical conference, and then attach call with Playback() to the same conference. For now, the Redirect() part is working fine, but due to lack of time, i haven't got further. On Monday 08 October 2007 14:13:38 Jaswinder Singh wrote: See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk to one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . But i dont know how to play a recorded file in it . My collegues tried this but unsuccessfully. The basic idea is to use local channels - one is bridged to Chanspy() and second to Playback(). I'm not sure what is the problem, but theoretically also this should work. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Manager API ! (System) command
On Wednesday 10 October 2007 07:04:02 robert home wrote: I need to issue some system commands via the Asterisk manager API. From the CLI the ! (system command) works fine, but when connected via the manager API it fails. Does anyone know why, or of a work around? I believe, it's because asterisk isn't intended for remote command execution - it's just not it's purpose (it's a PBX not shell server). I suppose the code of handling ! is in client part of asterisk CLI, not server. There are other far much superior and faster ways how to do that. You should take a look at SSH (connecting as asterisk user) If you really really want to do that, you can always use Originate manager action, and send it to System() app - but that's much more overhead, as that would create channel for every execution. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
At this point I was wondering if Asterisk gets real benefits on systems with several cores (up to 8 in Dell PE2950) for a system that will handle up to 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax (Sangoma A400D PCI card). I suppose that yes. Asterisk uses pthread, and it should distribute load across multiple cores. However, i doubt that you will need that much for 35 simultenous calls. I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
On 10/14/07, Andrew Nowrot [EMAIL PROTECTED] wrote: Hi I have a question if there was a major change in CDR? Few days ago I have upgraded to 1.4.12.1 from 1.4.4 and something bizarre happened. After the upgrade I have no call details in the cdr table when the call did not go through because of for example: Unable to create the channel of type Sip - no route to destination. In such situation the call does not exist in the cdr table while it was there when the same situation happened in 1.4.4. I also have this message in the console when an outgoing, noanswered call terminates: cdr.c:434 ast_cdr_free: CDR on channel 'Zap/2-1' not posted What cause this behavior? Is it a bug or misconfiguration. I tried to google this issue but unfortunately it does not reveal anything useful. Yes, there's a change. For me it's completely unacceptable, so i reverted the patch (http://bugs.digium.com/view.php?id=10659). The thing is that one-channel CDRs without answer are not written. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PSTN failover
On Tuesday 16 October 2007 06:21:45 Andrew Kohlsmith wrote:
GotoIf($[${DIALSTATUS} = BUSY]?busy)
GotoIf($[${DIALSTATUS} = NOANSWER]?noanswer)
GotoIf($[${DIALSTATUS} = ANSWERED]?answered)
Dial(Zap/...)
Of course, I do this inside a macro, and I emit correct CDR and correct
hangupcauses for those who use my system.
Dialing twice like that without checking your return value is an invitation
for future problems.
Well, as far as i have tried - i never get ANSWERED in DIALSTATUS. Only thing
that continues is h extension.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR
On 10/17/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 16 October 2007 15:25:13 Philipp Kempgen wrote:
Michael Collins wrote:
I don't know if it's relevant or not, but I do know that at least one
legacy PBX vendor (NEC) has a 'solution' that helps with some of the
sillier CDR's that could get generated. They have what they call a
pseudo-answer timer which is basically just a way of saying, If a
call doesn't last for at least X number of seconds then it really isn't
a call and no CDR should be generated. It is a bit of a case of
throwing away all really short phone calls, even legit ones, but it does
also get rid of the silly stuff: I pick up, get dial tone, then hang up
or I pick up, dial ext 1234, let it ring for two seconds and then hang
up.
I would definitely want a CDR record in this case.
Out of curiosity... why? Both you and Atis seem to want to see CDRs for
non-calls, and I'm unable to see why.
Hmm, i wanted to post a link on my mail to -dev list, but it's not
delivered there.. Ok, i'll paste it from my sent box:
--
The problem is that i don't get CDRs from IVR and from queue (if nobody
answered). I'm quite certain that the options for not-posting CDR should be
extended. For example if i'm setting userfield in dialplan - i might want
that CDR. Also, if i'm calling ResetCDR(w) i would really want that CDR (and
probably also next one).
I suppose, i could put Answer() in IVR, but i don't want it there, as nobody
actually picked that up (easier for accounting). And i defineately don't want
to do Answer() in queue's CDR (i want to know if somebody answered customer)
Can you please explain what was the cause of those fake CDRs? As i have
noticed - they couldn't be controlled from dialplan (no userfields, etc).
It's a good thing that they got removed, but those rules should be smarter.
Regards,
Atis
/* My dialplan */
context external {
1234 = {
Set(CDR(userfield)=incoming)
Answer()
goto ivr|1|1
}
}
context ivr {
1= {
ResetCDR(w)
Set(CDR(userfield)=ivr)
...
goto queue|2|1
}
}
context queue {
2 = {
ResetCDR(w)
Set(CDR(userfield)=queue)
Queue(2);
}
}
context agent {
_X. = {
Set(CDR(userfield)=agent)
Dial(SIP/${EXTEN});
}
}
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem: features (from features.conf) not available if call was originated by manager API or call file
On Sunday 14 October 2007 15:02:43 Volker Sauer wrote:
Hello asterisk-users,
I setup my asterisk to support several features like
automon,blindxfer,atxfer,parkcall etc. by using features.conf and the
global variable
DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in
extension.conf. Every Dial() command in my diaplan has the appropriate
parameters out of {tTkWwW}.
For calls from my SIP phones everything works fine. Pressing #1 will
transfer, pressing ## will automon and so on.
But if I originate the call from either a call file (which is used by a
callback application on my setup) or the manager api (which is used by a
webadressbook, which automatically dials and connects to the phone on
your desk) these features are not available even the manager api jumps
to the same context as a normal call from a phone and uses the same
Dial() command with {tTwWkK}. This means, that alle the nice feature
keys which normally work do not if I originate the call over the
webfrontend.
Therefore, either the global variable is not known to the call which
came from the call file or the manager api or something else is going
wrong.
If it's a global variable, and it works one way, but not another - then you
should post a bug.
You can also try using in your call file:
Set: DYNAMIC_FEATURES=automon#...
If this doesn't help either, you can dial to Local channel, and there execute
a Dial(), and set variables if necessary.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play sound on hangup
Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play sound on hangup
On Wednesday 17 October 2007 19:03:55 Tony Mountifield wrote: Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? So A calls B, who answers; A talks to B and B then hangs up - you want a sound file played to A? Assuming A is executing in the dialplan, and calls Dial() to call B, you might be able to do it by giving the 'g' option to Dial(), so that the calling channel continues after Dial has returned. You then follow the Dial with a Playback() command. However, in the above scenario, if it is A who hangs up, I don't think you can play a message to B, since B is not executing in the dialplan. Ok, i thing - this better explains my needs - i need to play sound to B, whenever A hangs up. Actually it's executing dialplan part with h extension - but i haven't been successful in playing sound there. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play sound on hangup
On Wednesday 17 October 2007 18:54:58 Philipp Kempgen wrote:
Atis Lezdins wrote:
Does anybody have some ideas - how to play a sound file on channel, after
that bridged channel got hanged up?
---cut---
g- Proceed with dialplan execution at the current extension if the
destination channel hangs up.
---cut---
Maybe something like this would do it:
Dial(tech/123,,g);
if (${DIALSTATUS} = ANSWER) {
Playback(somefile);
}
Well, as i have tried - i never get ANSWER status there - i just got thrown to
h extension.
Plus - i want to do it on other half of channel - i have incoming calls routed
trough queue (and Agent/ channels), then Dial()'ed to local SIP. In case when
caller hangs up - i would like to play the sound.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk using 200% CPU and then crashing...
On Wednesday 17 October 2007 19:09:23 Carlos Chavez wrote: Why would inserting a multiport card affect Asterisk and the server? How can I debug this situation? I do not have enough slots to insert three single cards of the same type so I need the multiport card to work. When Asterisk goes above 100% it will start ignoring commands you issue on the CLI but calls keep coming and going. At this moment I have to watch the server all day and if it goes above 100% I restart Asterisk when we have a low call volume. Obviously I cannot keep doing this forever. I recently had the same problems. Just that wasnt't related to installing any new hardware. You can check out the issue http://bugs.digium.com/view.php?id=10775 Could you provide your OS and glibc version? Also - can you try to disable IAX? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preflight check / lint
On Wednesday 17 October 2007 22:23:23 Andy Davidson wrote: Hi, Am writing scripts to manage configuration management and Asterisk. I would like to be able to point the asterisk binary at config directory with an asterisk.conf in it, and for asterisk to run a pre- flight check. A bit like a pint check in php, 'apachectl configtest' and lots of other tools. asterisk will then exit with 0 on a safe config, and 1 on a bad config. I can reject bad config and stop my config management script in the event of an error. Looking at the man page, it looks like this feature is missing. Anyone got another tool which can do this instead ? I know only of aelparse for extensions.ael (utils dir if i'm not mistaken) For the rest - you can get trough with simple regexps (config file formats are nearly identical). extensions.conf is generally the same format as other config files, however you should check syntax of applications, and priority ordering (if you are using it at all). Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parse error in GosubIf
On Wednesday 17 October 2007 22:57:41 Michael Iedema wrote: Greetings everyone, today I spent the last part of my day trying to find a parse error inside this snip: http://pastebin.ca/740081 If there's anyone who can shed some light on why my GosubIf condition is throwing a parse error, I'd greatly appreciate your insight. This was really frustrating and is probably a stupid mistake. Try removing spaces around = Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parse error in GosubIf
On Wednesday 17 October 2007 23:41:28 Michael Iedema wrote:
On 10/17/07, Ira [EMAIL PROTECTED] wrote:
At 12:57 PM 10/17/2007, you wrote:
If there's anyone who can shed some light on why my GosubIf condition
is throwing a parse error, I'd greatly appreciate your insight. This
was really frustrating and is probably a stupid mistake.
exten = h,n,GosubIf($[${SENDNOTIFICATIONS} = 1]?notify,1)
I just tried this again with quotes as many have suggested (thanks!)
but am still getting the same error. The first step in that
email-hungup macro is being executed and nothing else.
Also, this:
Oct 17 22:14:54 -- Executing [EMAIL PROTECTED]:2] GosubIf(Zap/3-1,
1?notify|1) in new stack
This means, the variable evaluates to 1 - only values are shown in log.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Crash related to asterisk -rx ?
On Thursday 18 October 2007 04:47:14 Jean-Denis Girard wrote: Hi list, Last Friday, an Asterisk server became unresponsive after ~8,5 months of smooth operation (~32 calls). Server did reply to pings, but no ssh, no more console login. Also Asterisk no longer took calls, but ISDNguard watchdog was still alive. Looking at the logs after reboot, I could not find anything significant, except in a file created by the following command via a cron job: date /var/log/asterisk/calls.log ; asterisk -rx show channels concise /var/log/asterisk/calls.log Two days before the crash, the calls.log file started to be filled with the Asterisk console messages. I suspect this is what caused the server crash. Anybody seen this before, is this a known problem with asterisk -rx commands? Yup, it's also a problem for me, but it haven't ever crashed server. It just makes specific remote process unresponsive. There's a patch for 1.4, but i guess it wouldn't be hard to backport it for 1.2 http://bugs.digium.com/view.php?id=10847 you might also want the one mentioned in comments: http://bugs.digium.com/view.php?id=10888 Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip reload causes unreachable
On Thursday 25 October 2007 15:36:44 Admin DeryTelecom wrote: Hi I have a asterisk with many phones (type=friend) When I issue the command sip reload some of the phones become unreachable and they come back just after. I guess that the sip.conf file is too big and asterisk takes too much time reloading the entire file. Is there a way to avoid this probleme or another way to add/remove sip phones dynamically ? Realtime? http://www.voip-info.org/wiki-Asterisk+RealTime+Sip Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] replace astdb with a cluster-capable sql database engine
On 3/8/08, Vieri [EMAIL PROTECTED] wrote: I've been searching the Internet for information regarding the replacement of astdb with a modern sql engine. There are several reasons one would like to do this. First of all, external applications have a hard time reading/writing to the now-old astdb format. Also (and this is what interests me most), the sql astdb could easily be clustered throughout several servers (I'm looking for a master-master MySQL 2-server cluster solution). Asterisk has brought up Realtime which is very powerful but, correct me if I'm wrong, it still requires astdb internally. In other words, if I call Set(DB) in the dialplan then it will always be using astdb regardless of realtime. Some projects like Callweaver have forked from Asterisk 1.2 and replaced astdb with sqlite. I'm wondering if Asterisk has plans to allow the user to choose the astdb backend: standard db1, sqlite, MySQL (which I would use with nbcluster for my clustering purposes), Postgresql with Slony-II, PGcluster, etc. Or is it already possible? There has been some talk on this before: http://lists.digium.com/pipermail/asterisk-dev/2004-December/007846.html Also, the func_odbc feature seems to be very powerful: http://www.asteriskpbx.org/func_odbc but: 1) would there be potential issues with db handles on a very busy asterisk system after a relatively long run time? 2) would there be a way to map the odbc function(s) to the DB functions (Set(DB), read and write, DBdel, etc) so that rewriting the whole dialplan would not be necessary? (that's the whole point of defining a different astdb backend) If there are known problems/issues/projects/alternatives then please let me know. There are really not much dependencies in asterisk to AstDB. As i recall - some SIP registration data is stored in AstDB, and persistent queue members (but you can replace that with Realtime queue members). For your own custom data you can use Realtime engine - it has INSERT and DELETE support in 1.6, and it's easily backportable to 1.4 (if you're interested i can give you working patches). All you have to do is declare realtime class in extconfig.conf, and then use Set(REALTIME()=...). For more info on this see http://www.voip-info.org/wiki/index.php?page=Asterisk+func+realtime Also there's a mysql command in asterisk-addons, but you have keep track of connections - connect and disconnect in dialplan. Or there's odbc module that creates permanent connection, and allows you to declare SQL functions with replacable variables, but personally i don't like having additional layer. For this you can search mailinglist, it's been described numerous times. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10
On 3/9/08, Godwin Stewart Horwich IT Services [EMAIL PROTECTED] wrote: On Fri, 7 Mar 2008 16:08:31 +0200, Mindaugas Kezys [EMAIL PROTECTED] wrote: Then you can change channel language in front of VoiceMail() app and in appropriate place put auth-thankyou file which is recorded/made by you. Much as I dislike this kludge because of the potential for b0rkage when Asterisk is updated, for now I've backed up the original auth-thankyou.gsm and symlinked silence/1.gsm to auth-thankyou.gsm. I think that giving 's' argument should silence all prompts including auth-thankyou. You should report a bug on http://bugs.digium.com , fixing this should be trivial. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Silencing VoiceMail() app in * 1.4.10
On 3/9/08, Godwin Stewart Horwich IT Services [EMAIL PROTECTED] wrote:
On Sun, 9 Mar 2008 16:22:35 +0200, Atis Lezdins [EMAIL PROTECTED] wrote:
I think that giving 's' argument should silence all prompts including
auth-thankyou. You should report a bug on http://bugs.digium.com ,
fixing this should be trivial.
It isn't that trivial.
I've looked at the source and the silent flag is not passed all the way
down the chain to the function that actually does the recording.
In apps/app_voicemail.c, the option is parsed by vm_exec() and passed on to
leave_voicemail().
leave_voicemail(), however, doesn't pass it down to play_record_review().
So by the time the call stack goes through ast_play_and_record_full() and
__ast_play_and_record() in main/app.c, where we see the foillowing code,
the status of the silent option is long lost:
if (outmsg == 2) {
ast_stream_and_wait(chan, auth-thankyou, chan-language,);
}
I will, however, work on a patch to pass the silent option down the chain
to this function, but it's going to mean a major overhaul.
Maybe you should ask for best way for this in asterisk-dev. I checked
wat you're saying and it seems to me that more logical would be to
play auth-thankyou in application, not __ast_play_and_record(), but
it may break some concept.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audiocodes MP124-FXS replying BUSY when line is not.
:14] VERBOSE[31897] logger.c: Transmitting (NAT) to ee.ff.gg.hh:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK3977e3c7;rport From: 28901-2067217913 sip:[EMAIL PROTECTED];tag=as18481a04 To: sip:[EMAIL PROTECTED];tag=1c1673732975 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- [Mar 10 11:13:14] VERBOSE[30165] logger.c: -- SIP/90166-c45079a0 is busy [Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: Everything is fine. [Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: Delete SQL: DELETE FROM channels WHERE uniqueid = '1205172794.6453' [Mar 10 11:13:14] DEBUG[30165] res_config_mysql.c: MySQL RealTime: Deleted 1 rows on table: channels [Mar 10 11:13:14] DEBUG[30165] chan_sip.c: Call to peer '90166' removed from call limit 8 [Mar 10 11:13:14] VERBOSE[30165] logger.c: == Everyone is busy/congested at this time (1:1/0/0) - end of log --- -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call tracing - Asterisk 1.4
On 3/11/08, Louwrens Benadé [EMAIL PROTECTED] wrote: Hi guys I've just read this about the upcoming release of * 1.6: Better reporting through a new call event logging capability in Asterisk 1.6 will allow complete tracking of events that take place during a call. The goal, according to Fleming, is to provide more detail than traditional CDR (Call Detail Recording) features offer and to allow for more granular tracking and auditing. That sounds brilliant! But I'm in desperate need of something to handle call tracing in 1.4... Does anyone know how I can accomplish this? I thought about using the originating uniqueid and populate for every event related to the call (transfers, etc), but I'm having trouble reading the dialplan to see what executes where :( That's the way how i have it workin. Of course, this wasn't done in one day, i've been working on details for weeks. Generally i use CDR, and manipulate it with ResetCDR, NoCDR, and link them together by first uniqueid. This works great for IVRs, extension2extension calls, outgoing calls, blind transfers, queues. So i can take any call and see what was done to it, where it was transferred, duration of each step and so on, so on. However it won't work for conferences (you don't know that person will join conference unless it joins, and then it's too late to change uniqueid, first cdr may be already writted), and i haven't implemented that for blind transfers. But generally if you want all that in DB, manipulating CDR is the way to go. When you will have more specific questions, please ask, i'm sure somebody will answer :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call tracing - Asterisk 1.4
On 3/12/08, Louwrens Benadé [EMAIL PROTECTED] wrote:
That's the way how i have it workin. Of course, this wasn't done in
one day, i've been working on details for weeks. Generally i use
CDR,
and manipulate it with ResetCDR, NoCDR, and link them together by
first uniqueid. This works great for IVRs, extension2extension
calls,
outgoing calls, blind transfers, queues. So i can take any call and
see what was done to it, where it was transferred, duration of each
step and so on, so on. However it won't work for conferences (you
don't know that person will join conference unless it joins, and
then
it's too late to change uniqueid, first cdr may be already writted),
and i haven't implemented that for blind transfers. But generally if
you want all that in DB, manipulating CDR is the way to go.
When you will have more specific questions, please ask, i'm sure
somebody will answer :)
So I'm not the only one :)
Ok, because of my lack of knowledge about using the dial-plan, I've resorted
to using Trixbox (don't laugh). I've managed to find where the initial
uniqueid is inserted which I then pump into a variable, and from there into
the 'userfield' in the CDR.
The problem I'm having at the moment is that I can't figure out when the
next hit in the CDR takes place. I've found the macro that (I think)
generates it, but no matter what I try, I can't populate the 'userfield' for
the next event.
So here are my questions:
1. Is the next event in the CDR inserted by ResetCDR or NoCDR?
NoCDR wouldn't cause that, as that's supposed to skip posting current
CDR. Next entry would be caused by either ResetCDR(w) or some
application that creates new channel (i.e. Dial or Queue). You can
enable full log and set verbosity and debug to higher values, to see
all what's going on.
2. Can I use a locally defined variable ( exten =
s,n,Set(v_identme=${CDR(UNIQUEID)})) ) or do I have to use a global
variable?
I'm not sure about value of ${CDR(UNIQUEID)}, but you can use just
${UNIQUEID}. If you want to pass variable to child channels, you
should make it inheritable. I'm using:
Set(__call_id=${UNIQUEID})
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue log vs. cdr
On 3/13/08, Vieri [EMAIL PROTECTED] wrote: Hi, Surely, I must be overlooking something. If I run the following SQL queries I don't get the same number of rows. Is this coherent? mysql select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 2008030800 and 20080313145900 group by callid; 357 rows in set (0.01 sec) mysql select * from cdr where dst = 4010 and calldate between 2008030800 and 20080313145900 group by uniqueid; 219 rows in set (0.19 sec) Thanks! Hmm, didn't knew that queue_log can be written into MySQL.. that's something useful for me :) Is callid in queue_log the same uniqueid? You can do something like this: CREATE TEMPORARY TABLE a TYPE=MEMORY select * from queue_log where queuename = '4010' and FROM_UNIXTIME(time) between 2008030800 and 20080313145900 group by callid; CREATE TEMPORARY TABLE b TYPE=MEMORY select * from cdr where dst = 4010 and calldate between 2008030800 and 20080313145900 group by uniqueid; and then compare: SELECT * FROM a WHERE callid NOT IN (SELECT uniqueid FROM b) SELECT * FROM b WHERE uniqueid NOT IN (SELECT callid FROM a) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update_call_counter: Call to peer '2509' rejected due to usage limit of 1?
On 3/17/08, Rajkumar S [EMAIL PROTECTED] wrote: On Mon, Mar 17, 2008 at 6:30 PM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: Forgot to add: Multiple queues fo sip phone, it is normal that sometimes it is ringed, as reported busy for 1 queue and free for another. you limitited incoming call to max 1 ' incominglimit=1' so ;) My understanding was that if a SIP phone is busy, either due to a call from queue or a call from another sip phone or even making an out bound call, the queue application would detect that and skip trying that channel. Is this assumption wrong ? If that would be queue, it would have different log entry. This seems, a result from Dial(SIP/2505,,). There are two different settings. You can increase call-limit (or incominglimit) in sip.conf - so devices will be able to take several simultenous calls. So, even if SIP device has one call (and call-limit is more than one), device state of SIP device will be In Use, and that's where ringinuse parameter of Queue application comes in - if set to 0, Queue won't ring and you will see a bit different message. Hope that this explains architecture. As for current problem - i suspect that device state don't get updated correctly for Queue application, so Queue tries to dial device, and call-limit blocks it from doing so. There's a patch, currently in testing (issue 12127), it should fix this, however if you intend to keep incominglimit to 1, and don't use local channels - there's nothing to worry about. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Order of queue member list
On 3/17/08, C. Chad Wallace [EMAIL PROTECTED] wrote: We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed a change in the behaviour of the queues--a change that we cannot live with. We've used AddQueueMember/RemoveQueueMember to manage logging into and out of our queues for over a year now with Asterisk 1.2, and in that version the queue members were sorted in such a way that the person who had been logged in the longest would be the first one to get a call. But when we deployed 1.4 last week, we noticed that the member list was no longer sorted based on login time. It seemed to have a pre-set order that members were always placed into. After looking at the code (apps/app_queue.c), I found the cause of this. In 1.2, the members were stored in a linked list, so when someone logged in, they were placed at the end, and when calls were handed out, it was done starting at the front of the member list (or vice-versa, but either way, it has the same effect). In 1.4, the member list was changed to an ao2_container, which apparently uses a hash table, and iterates through the list in a fixed order, meaning one of our agents is always the favourite for a call, and it is quite unfair. Now, I know that the ordering of members in the queue in 1.2 was not documented, and it may not have even been intentional, but it was very appropriate for our business model, and we'd like to find a way to get it back. Is there a way to control the order in which the ao2_iterator returns the items? Even a random distribution would be better than the current--which always favours some agents over others. And before anyone mentions the strategy setting in queues.conf, I should say that we use leastrecent, but because of the ratio of agents to queues in our business, the strategy doesn't come into effect immediately. With many agents answering each queue, it takes a while for each of them to get a call. Until then (which usually takes about half of each day), the calls are distributed based on the ordering of the member list. We have switched to the rrmemory strategy for now, but we've yet to notice what effect that has--and our ideal would be to use leastrecent along with the behaviour that Asterisk 1.2 exhibited. I would suggest adding: cur-lastcall = time(NULL); within create_queue_member() function. This will allow you speed bonus from hashtable in some places, and will make sure the login time gets registred. You can also consider updating lastcall in set_member_paused() - i'm having both of those. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 reliability problems
I would suggest taking latest 1.4 branch from SVN (or 1.4.19-rc3 when it's out). There has been few deadlocks fixed since rc2. Recompile asterisk with DEBUG_THREADS enabled (in make menuselect), If you're not using safe_asterisk script to start it, you should execute also ulimit -c unlimited before launching asterisk.. When your asterisk is deadlocked, open CLI and execute core show locks. Copy that output, and submit to bugs.digium.com - it will tell developers where exactly is problem. Then, do killall -11 asterisk. It will dump asterisk to core file, and that might provide helpful information later. If your have been requested backtraces, look in /tmp (or in directory you launched asterisk from) for core file. Open that core file with gdb /usr/sbin/asterisk core. and take a dump of thread apply all bt full (make sure you set set pagination off in gdb before this) Regards, Atis On 3/18/08, Norman Franke [EMAIL PROTECTED] wrote: Check around on bugs.digium.com. You'll find a number of issues reported that sound similar. I'm hoping that 1.4.19 will fix a lot of stuff, since the release candidates seem much more stable to me. I couldn't keep Asterisk up for more than a few days before on 1.4.18. I've also applied a few SIP-related patches from various bug reports and things are much, much more stable. 1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many issues. Norman Franke Answering Service for Directors, Inc. www.myasd.com On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED] wrote: We have been experiencing some ongoing reliability problems with Asterisk for quite some time, and I am trying to find out if anyone else has experienced the same problems. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ztdummy problem causing playback () to fail
On 3/18/08, Pete Kay [EMAIL PROTECTED] wrote: Hi, I am having problem with my Asterisk installation and find out it has to do with ztdummy. if the ztdummy module is loaded, the asterisk playback() command will not play files. DTMF is still properly received. If the ztdummy module is unloaded, sound playback works again. Here is my version zaptel-1.4.9.2 linux-source-2.6.18 asterisk-1.4.18 Can anyone tell me how to fix it? Or should I just have ztdummy removed forever and the system will work? I saw from manual that ztdummy is required. ztdummy is required by meetme application. If you have no intention to use it, you might very well remove. I've seen this problem once, however recompiling everything and restarting helped me. I would suggest you just doing make clean on zaptel and asterisk, then compile first zaptel, then asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk.conf uniquename or sysname for uniqueid field in CDR
On 3/18/08, Vieri [EMAIL PROTECTED] wrote: --- Vieri [EMAIL PROTECTED] wrote: I set uniquename = MYHOST in asterisk.conf (under [options]) so that my uniqueid data shows up as MYHOST.time.seq. First of all, I would like to know if uniquename (or sysname?) will still be valid across future * versions (mainly 1.6). Secondly, is there a way to specify uniquename as an asterisk option at the command line? (asterisk -h doesn't show me anything regarding this feature) Finally, how can I set uniquename to a system value (say, dynamically set to whatever `hostname` yields)? Something like uniquename = `hostname` so that I don't have to statically set it on each asterisk server? I just realized that uniquename is only available after applying the BRISTUFF patches. So let me rephrase my question: will Asterisk ever include the uniquename feature in its base code? If not, why? (I would prefer not to apply BRIstuff since I don't have Junghanns hardware). Look into doc/asterisk-conf.txt - probably you can use systemname. Asterisk config files also support #exec directive, so you can create your regular asterisk.conf without sysname and create shell script: #!/bin/bash cat asterisk.conf.template echo sysname=`hostname`. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling 3 different call ending causes
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply the
${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the instructions
to continue executing to the next priority for that extension, whereas if
the call completes (Dial() is successfully connected), this does not happen.
I''ve tried to use the h extension in combination with the ${CHANNEL} in the
dialplan as suggested on the wiki page, but I haven't had any luck with it.
For this test I have a Sipura phone with number 1003 and a X-lite with 1203.
If I let the time go by (the 20 seconds defined in the Dial Command) I get
the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup
is
SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup is
SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel hungup is
SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small context with
the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = s,2,NoOp(Callee hangup)
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
This should allow you to distinguish caller and callee hangups. I
suppose dial time limit will match Callee hangup, but you can check
that by ${ANSWEREDTIME} or some sort of timestamp checking before and
after Dial (altough that would include ringing time)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Handling 3 different call ending causes
On 3/20/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Date: Wed, 19 Mar 2008 11:31:57 +0200
From: Atis Lezdins [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Handling 3 different call ending causes
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1
On 3/17/08, Tobias Ahlander [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
Hello List,
I'm using a dialstring like the one below. I want to have three
different things happening depending on exit cause.
Dial(SIP/${phonenumber},20,gL(2[:5000][:5000]))
These 3 things could happen:
1, Caller hangs up
2, Callee hangs up
3, The 20 seconds is up and call is terminated from Asterisk.
Is there a way to separate these 3?
You can handle the 'h' extension in the dial plan, which will supply
the
${CHANNEL} that was hung up, and possibly some additional dial plan
variables as well:
http://www.voip-info.org/wiki/index.php?page=Asterisk+h+extension
Using these, you can piece together who hung up on whom, etc.
#2 is handled by fallthrough in the dial plan that causes the
instructions
to continue executing to the next priority for that extension, whereas
if the call completes (Dial() is successfully connected), this does not
happen.
I''ve tried to use the h extension in combination with the ${CHANNEL}
in the dialplan as suggested on the wiki page, but I haven't had any luck
with it.
For this test I have a Sipura phone with number 1003 and a X-lite with
1203.
If I let the time go by (the 20 seconds defined in the Dial Command) I
get the following:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
If I let the Sipura hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Lastly if I let the X-lite hang up I get:
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/1003-08a491b8, Channel
hungup is
SIP/1003-08a491b8) in new stack
Yes they are all the same :(
Perhaps there's something wrong with my code? Its just a small
context with the following for this test:
[hangupcause]
exten = s,1,Dial(SIP/1203,30,gL(1[:5000][:5000]))
exten = s,2,NoOp(Callee hangup)
exten = h,1,NoOp(Channel hungup is ${CHANNEL})
Have I missed something basic here or what?
This should allow you to distinguish caller and callee hangups. I suppose
dial time limit will match Callee hangup, but you can check that by
${ANSWEREDTIME} or some sort of timestamp checking before and after Dial
(altough that would include ringing time)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
Hello List,
Ok, I solved it by using this code. This will work for me since the variable
${timeleft} is always in complete seconds. Thank you all for the ideas and
pointers :)
context hangupcause {
s = {
Set(timeleft=7000);
Dial(SIP/1203,30,gL(${timeleft}[:4000][:4000]));
if(${timeleft} = (${ANSWEREDTIME}*1000)) {
jump [EMAIL PROTECTED];
} else {
jump [EMAIL PROTECTED];
}
}
h = {
NoOp(Caller Hangup);
}
}
context hangupcause2 {
s = {
NoOp(Callee Hangup);
}
}
context notimeleft {
s = {
NoOp(Time's up!);
}
}
I would change that to = just for reliability - you never know :)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
On 3/20/08, Stefan Schmidt [EMAIL PROTECTED] wrote: hello, i am trying to set up a asterisk server (version 1.2.26 by now) with realtime configuration but the user shouldnt register directly to the server, instead i have set up a ser registration proxy. Everything works fine so far, but i can´t use the hint feature. Its possible to subscribe to a given hint, but the status is allways unavailable and also i dont get a notify. Could someone help me finding a solution for this problem? I want to get notifies for hints where the user isnt registered on the asterisk itself. Thanks best regards Steve Smith ps: allready posted on Dev lists with the result this isnt a dev- related topic. What did you mean by realtime config? Realtime SIP users, realtime dialplan? If it's just SIP users, you should have some success with rtcachefriends=yes in sip.conf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] hint status unavailable
On 3/20/08, Steve Davies [EMAIL PROTECTED] wrote:
On 20/03/2008, Johansson Olle E [EMAIL PROTECTED] wrote:
20 mar 2008 kl. 09.32 skrev Stefan Schmidt:
hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything
works
fine so far, but i can´t use the hint feature. Its possible to
subscribe
to a given hint, but the status is allways unavailable and also i dont
get a notify.
Could someone help me finding a solution for this problem? I want to
get
notifies for hints where the user isnt registered on the asterisk
itself.
That is something we all want, but it doesn't work now unless you add
a third party software. I haven't seen anything that solves the issue,
but have
a few ideas.
The question here is how should one asterisk be able to know anything
about
devices it doesn't control? It's a pbx, not an artificial intelligence
software.
There is work going on in the development group to make it possible
to apply a message bus between Asterisk servers so that Asterisk
servers can share call states. When that is up and running and tested,
it will be part of a future Asterisk release.
So the answer in short is not possible today, maybe tomorrow
Regards,
/olle
Perhaps in a similar thread, is it possible to somehow SET the state
of a hint from the dialplan? Perhaps a bit like:
Set(${ChanIsAvail(hint,234)}=Busy)
or perhaps have a pseudo-device facility where you can add it to the
end of the hint list to hint-the-hint. Something like:
exten = 234,hint,SIP/myphonePSEUDO/234
exten = *78,1,ChanAvailIs(PSEUDO/234,Busy)
exten = *791,ChanAvailIs(PSEUDO/234,Unknown)
This could be very useful for presence indication.
Huh, this hint hint would be useful for queues with local channel
state_interface too.. i think some general usage way could be added to
allow combining of device states.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] callers in queue passed to agents who accept only one call at a time
On Thu, Mar 27, 2008 at 6:32 PM, Vieri [EMAIL PROTECTED] wrote: I have a queue I configured as strict and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy). Instead, it's as if they had call waiting and incoming calls keep popping up while they're conversating with the first caller and they would like to avoid this. I guess the easiest solution would be to find an open-source or free softphone that can be configured to accept only one call at a time (currently using SJphone). Another solution would be if I could tell the Queue() application that if an agent is InUse then don't pass the call. Still another yet more delicate solution would be to have a custom script receive manager events related to the queue which in turn replies with an agi command. For example, whenever an agent answers a call I think that an event such as QueueMemberStatus can be triggered (although I don't know how). If the custom script could receive this event in realtime then it would run an agi command such as QueueRemove(busyagent...). When the agent is free again I suppose the same event is triggered and the custom script can QueueAdd(freeagent...). Could anyone please give me some pointers on this? In queues.conf set ringinuse=no Also make sure that you don't use realtime sip peers (or use rtcachefriends with that). Probably you also need call-limit set to any value in sip.conf For more info see http://www.voip-info.org/wiki-Asterisk+config+sip.conf Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interrupting MOH
Sorry for top-posting, but seems everyone on this thread did so. Also that would be my suggestion for now - call queue with periodic-announce. However i see that this would make nice architectural improvement - allow inject sound files into MoH stream. This would be useful for example in call queues - to inject all the queue announcements into MoH directly, rather than play them while blocking further queue actions. Regards, Atis On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: I think that's still a better idea than using a dump the caller into meetme hack and is actually what I was going to suggest. If you want something simpler than a queue then inject the sounds into the moh already. On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote: You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interrupting MOH
On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson [EMAIL PROTECTED] wrote: You could also, conceivably, handle this outside of asterisk by using a more complex MOH stream source. For instance, use a shoutcast client as the MOH source, run your own shoutcast server streaming your music and have a script set up to periodically interrupt the stream being served to the shoutcast server and inject an announcement. (Keep in mind that this is an off the top of my head suggestion so I don't have exact details for implementation, but I'm sure it can be done.) That would need one shoutcast stream per call.. not very reasonable.. Regards, Atis Good luck, Brent Matt Florell wrote: Hello, We achieve this using an AGI script in the VICIDIAL project for our version of inbound queues. You start MoH then when you stream a sound to the channel it will stop MoH then after the sound is done you start MoH back up again. Probably a bit more involved than what you want, but it dose work well for us. MATT--- On 4/2/08, Atis Lezdins [EMAIL PROTECTED] wrote: Sorry for top-posting, but seems everyone on this thread did so. Also that would be my suggestion for now - call queue with periodic-announce. However i see that this would make nice architectural improvement - allow inject sound files into MoH stream. This would be useful for example in call queues - to inject all the queue announcements into MoH directly, rather than play them while blocking further queue actions. Regards, Atis On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge [EMAIL PROTECTED] wrote: I think that's still a better idea than using a dump the caller into meetme hack and is actually what I was going to suggest. If you want something simpler than a queue then inject the sounds into the moh already. On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis [EMAIL PROTECTED] wrote: You may be able to achieve the desired result using queues rather than Dial statements. Overkill perhaps, but it's the only way I can think to implement it at the moment. John Millican wrote: Tilghman Lesher wrote: On Tuesday 01 April 2008 05:14:25 Pete Kay wrote: I am hoping someone can help me out on this. I want to be able to interrupt MOH every X seconds after the DIAL command is executed. The interrupt greeting is something like please wait while we transfer your call. How can I do that? Within the DIAL options, I can't see any announce frequency or options that can help. Could anyone please tell me how that function can be accomplished? The only way to do that currently is to implement the prompt within the MOH stream itself. Just off the top-o-my head(YMMV), couldn't you create a meetme and play hold music into the meetme and then also play the prompt into the meetme at the same time without interrupting the hold music? This would obviously not work for high load but... JohnM ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689
Re: [asterisk-users] Asterisk 1.4.19 crash with Realtime using SIP peers
On Wed, Apr 9, 2008 at 5:29 PM, Trevor Peirce [EMAIL PROTECTED] wrote: Mindaugas Kezys wrote: Hello, Asterisk 1.4.19 crashes everytime using Realtime and SIP peers Yes I also saw this and had to revert. Calls to the IVR seemed to be fine, but as soon as two peers call each other it crashes as the call progresses (never connects). I haven't had a chance to explore any further and therefore haven't posted a bug either. Perhaps this weekend if nobody does first. So far works fine for me. Sample peer setup below. Had one issue with peers where ipaddr was 0 (and hostname used instead), but adding this patch ( http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?r1=113012r2=113240 ) seems to solve everything. Regards, Atis *** 1. row *** id: 2 name: 21168 accountcode: NULL amaflags: NULL callgroup: NULL callerid: Atis 21168 canreinvite: no context: default-sip defaultip: NULL dtmfmode: rfc2833 fromuser: NULL fromdomain: NULL fullcontact: sip:[EMAIL PROTECTED]:5061 host: dynamic insecure: NULL language: NULL mailbox: [EMAIL PROTECTED] md5secret: NULL nat: yes deny: NULL permit: NULL mask: NULL pickupgroup: NULL port: 5061 qualify: no restrictcid: NULL rtptimeout: NULL rtpholdtimeout: NULL secret: 21168 type: friend username: 21168 disallow: allow: all musiconhold: NULL regseconds: 1207763735 ipaddr: 192.168.1.123 regexten: cancallforward: yes setvar: call-limit: 4 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queue
Hey, I just found out today that it doesn't work on Asterisk 1.4.19 (at least for realtime queues) if you have autofill=yes in queues.conf. However it works if you add it in queue settings for each queue, for realtime that would be ALTER TABLE queue_table ADD COLUMN autofill TINYINT(1) UNSIGNED DEFAULT 1; For following this issue, see http://bugs.digium.com/view.php?id=12445 Regards, Atis On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Global call limit
On Wed, Apr 16, 2008 at 12:46 AM, Vinz486 [EMAIL PROTECTED] wrote:
Hi,
i'm new in asterisk programming.
Maybe my question was posted thousand times but i found nothing using google.
I'm looking for a method to limit the total simultaneous calls
(inbound and outbound) that pass from internal phones to 2 SIP
providers.
I found the calllimit option but it works only on a per-channel basis.
Instead i want limit the total amount of calls, abstracting from which
SIP provider us used.
This to keep good audio quality for active calls and rejecting new
arriving: this is needed for a PBX connectect with a poor ADSL having
only 256kbit in upload: so i want permit only 3 calls (256 / 80 kbit)
and rejecting the 4th.
Any solutions?
if (${GROUP_COUNT([EMAIL PROTECTED])}) function in combination with
Set(GROUP(a)=x)
or Set([EMAIL PROTECTED])
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license count...
The codec in use for a specific channel doesn't even care if that channel exists over zapata analog or digital cards, sip channels, iax[2] channels, smoke signals, etc. If you care to use ping pong balls and the atlantic ocean as your medium, you should be able to interface with the g729 codec if you still needed to :D Although I wouldn't expect there to be much error correction inherent in the Atlantic. I would not risk sending my data trough new cutting edge transports You mentioned. Instead I prefer to use proven technologies, and preferably documented in RFC - for example RFC 2549 IP over Avian Carriers with Quality of Service. There are even some modifications to this by using flash cards instead of paper, and that beats speed of ADSL. However that still doesn't seems best for my VoIP traffic because of latency. The codecs are modules for *asterisk* and not for the cards themselves. That's true. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
Hi, I experience my log flooded with warning messages like this: [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up I traced this down to point when we added to sip.conf status notifications: allowsubscribe=yes rtcachefriends=yes So, those notifications allow for queue to display (In Use) etc, and creates no warnings for other devices except Audiocodes gateway. I wonder is there any way how to disable this message in Asterisk, or make Audiocodes act correctly? Below is the sip debug for this (xx.xx.xx.xx is Audiocodes, yy.yy.yy.yy is Asterisk). Regards, Atis - [Apr 14 01:30:24] VERBOSE[19514] logger.c: Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: NOTIFY) [Apr 14 01:30:24] VERBOSE[19514] logger.c: Reliably Transmitting (NAT) to xx.xx.xx.xx:5060: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport From: Unknown sip:[EMAIL PROTECTED];tag=as436bf308 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 0/0 (0/0) --- [Apr 14 01:30:24] VERBOSE[19514] logger.c: --- SIP read from xx.xx.xx.xx:5060 --- SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK788fbefd;rport From: Unknown sip:[EMAIL PROTECTED];tag=as436bf308 To: sip:[EMAIL PROTECTED];tag=1c73477527 Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY Contact: sip:xx.xx.xx.xx Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Content-Length: 0 - [Apr 14 01:30:24] VERBOSE[19514] logger.c: --- (10 headers 0 lines) --- [Apr 14 01:30:24] WARNING[19514] chan_sip.c: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WARNING: Remote host can't match request NOTIFY to call on Audiocodes MP-124 FXS
On Tue, Apr 22, 2008 at 3:15 PM, Grey Man [EMAIL PROTECTED] wrote:
For blind transfers Asterisk will send the call back to the dial plan
and into the TRANSFER (I think, could be a different name) context if
it exists. Within that context you can access the channel that was
answered on the original call using ${DIALEDPEERNUMBER}.
Note that this mechanism cannot be use for attended transfers as they
are not sent back to the dial plan for processing.
I apologize, but I don't have any problems with transfers. The
warnings I get in log appears there even without any calls going on.
Maybe You replied to wrong topic?
Regards,
Atos
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
Queue will continue if called person hangs up (and there's no option).
If caller hangs up, call goes to h extension in same context. Just the
same way as Dial with 'g'. There's a change in 1.6 that allows called
channel to continue if caller hangs up, so probably something like
this could be applied also to Queue (or was that actually working with
using Local channels?).
Regards,
Atis
On Wed, Apr 23, 2008 at 7:13 PM, AnDY [EMAIL PROTECTED] wrote:
Thank you for your answer.
But the Dial command has a option 'g' which means that after succes will
proceed next priorities in the dialplan. Is there something also for
Queue() because according to manual there is no option for it. So I am
looking for some other solution.
Andy
Tony Mountifield napsal(a):
In article [EMAIL PROTECTED],
[EMAIL PROTECTED] wrote:
Hello everybody.
I was looking for the solution but nothing found. I have this in my
extensions.conf:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
But when the call is placed in the queue and somebody answer it, it will
throw an error:
== Spawn extension (default, 211, 4) exited non-zero on
'Local/[EMAIL PROTECTED],2'
And no other command in extensions is executed.
Any suggestions?
Queue() is like Dial(), in that if it succeeds in connecting to someone,
it will not return to the next priority in the dialplan. However, if you
define an 'h' extension, that will get executed when the call is complete:
exten = 233,1,SetAccount(queue1)
exten = 233,2,Queue(queue1|rn)
exten = 233,3,NoOp(${QUEUESTATUS})
exten = 233,4,NoOp(${DIALSTATUS})
exten = h,1,NoOp(${QUEUESTATUS})
exten = h,2,NoOp(${DIALSTATUS})
Cheers
Tony
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Next step in extensions.conf after answer the phone in Queue
Atis Lezdins wrote: Queue will continue if called person hangs up (and there's no option). If caller hangs up, call goes to h extension in same context. Just the same way as Dial with 'g'. There's a change in 1.6 that allows called channel to continue if caller hangs up, so probably something like this could be applied also to Queue (or was that actually working with using Local channels?). On Wed, Apr 23, 2008 at 8:18 PM, Al Baker [EMAIL PROTECTED] wrote: Why would you want a channel to continue after the caller has hung up. I clearly am missing something here because I can't see what good that would be. What do people do with this Continued Channel ? What is is used for ? How Does having it help you ? ??? To play something to called party. I'm not familiar with that feature too deep, but I guess it's not caller channel but called channel that's continued. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue callers
On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime queue callers
On Tue, Apr 29, 2008 at 1:22 PM, Vieri [EMAIL PROTECTED] wrote: --- Atis Lezdins [EMAIL PROTECTED] wrote: On Mon, Apr 28, 2008 at 8:34 PM, Vieri [EMAIL PROTECTED] wrote: How can I get a list of the callers within a specific queue at any given moment? I need to get the caller IDs of all active calls in a queue then send them out via a udp socket to a listening application on the network (the only data I need to send are two fields: current timestamp and caller id of active queue calls). I have almost all the elements to do this except the best method to retrieve all active caller ids from a given queue. I was wondering if someone already did this. I tried writing a script on the server which connects to the Manager API and receives queue events. I'm basically using the AgentCalled event but it seems clumsy to efficiently detect when the call has ended (connect or abandon) and thus update the remote UDP listening app. I also tried another way by guessing which calls are active via tailing and grepping /var/log/asterisk/queue_log. Finally, a third script method tried parsing the output of show queue right after Callers:. Maybe this is all I really need for my purposes (although less efficient and less real-time than the queue events method because I would need to periodically poll the whole queue statistics) but I only get the originating channel and the wait time. I would require correlating the data to the caller's ID. Has anyone already done something similar? A simple example/script/suggestion would be greatly appreciated. I'm not sure that this is what exactly You need, but I have a patch for app_queue that will store and update queue callers (as well as update lots of fields for queue members) in realtime mysql table. This allows to do many requests for current queue state simultenously, and moves load from asterisk to mysql (which can be on separate machine). So, generally to get active callers with all their callerid/channel info You will have to do just SELECT * FROM queue_callers. It's not very finalized, so I haven't yet posted that to Digium for inclusion in next asterisk versions, but I intend to do that in future. It's been working stable on our production for several months. If You're interested, please reply, and I'll try to separate that patch out from other our patches. Currently I have it updated for 1.4.19, but also have some version for 1.4.14 Thanks Atis. That patch sounds really neat. Hope it gets into * soon. Just a doubt: suppose the mysql daemon dies for some reason. Will the patched app_queue still handle calls and not hang? It should, as asterisk throws INSERTs, UPDATEs and DELETEs for changing data (queue callers and queue member status), plus it loads existing queue members trough SELECT (as it's now with realtime queue members, just some extra fields). So, I suppose if MySQL dies in middle of operation, SELECT should fail and Asterisk should just continue with what it has in memory. Btw, You should be able to also use static or dynamic queue members (not realtime) in combination with realtime queue calls. Btw, I never experienced that MySQL dies, it's more often that Asterisk dies. So, are You interested in applying this patch yourself? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote host can't match request NOTIFY???
On Thu, May 1, 2008 at 1:38 PM, Alan Lord [EMAIL PROTECTED] wrote: Grey Man wrote: On Thu, May 1, 2008 at 7:54 AM, Alan Lord [EMAIL PROTECTED] wrote: Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '[EMAIL PROTECTED]'. Giving up. snip / Asterisk does not correctly match SIP NOTIFY transactions in at least some cases. Your problem may be related to http://bugs.digium.com/view.php?id=11848. Regards, Greyman. Thanks for that. Not sure I understand it all. I am not actually doing anything when these messages appear. They occur pretty much every minute or so. With or without any calls... There was a post week ago (I was having the same problem). For me it was caused by AudioCodes not understanding voicemail notifications. So, first You can enable SIP debug to see what packets are causing this, and if it's voicemail notifications, turn them off in sip.conf (mailbox= line). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime status feature - user feedback needed
Hello users, I had developed several patches that allows to monitor current status of queues/channels in realtime db. For example specifying realtime family channels will make asterisk to keep current channel list in realtime database engine. The same would be for queue callers and queue members (already partially available in 1.4). However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. *** So, realtime status - what's this all about? Basically you get output of show channels, show queues, etc directly in Realtime table (Realtime = database engine system for Asterisk). So, Asterisk will automatically update database upon any changes of channels or queues. *** Why would You need that? In beginning I created this in order to deal with large amount of monitoring software. If there's lot of users monitoring status, some kind of cache should be put into place. With current Asterisk interfaces this would mean either inquiring current status or developing a daemon that follows up all events and collects them to keep current picture always ready. I just decided to move this layer to database engine, which deals really good with this stuff. *** Rapid development of monitoring tools What it takes to create custom monitoring tool now? AMI event listener? AJAX page that gets changes from built-in webserver? All this takes lot of time to learn and start using. Adding just few config lines in extconfig.conf would allow to automatically populate database with current status - so it's accessible easily from any programming language. All the info is just there, no need for processing or analyzing. *** Performance / Scalability Inquerying queue status means that there is lock put on queue list, all queues are traversed, information gathered and then returned. If lot of instances of monitoring software need to have this information, it's obvious that this would mean too much locks. So, as database update is thrown whenever some change is happening, it means that no additional locks are created for monitoring purposes. Transaction is sent to database engine, which keeps relatively small tables of current status. Then any number of clients can access data directly without any locking. Even 200 concurrent calls with 10 new calls per minute would still be a tiny load for MySQL. This can also be scaled by moving database to another machine, thus allowing more raw CPU usage for Asterisk. *** Development maintenance Those changes doesn't introduce any new functions in asterisk code. They utilize currently available Realtime engine which is meant for storage of configuration data. This just extends use of this engine also for status data, so maintenance of this interface should not take lot of work. *** Current patches If You are interested in using those changes right away, here are some backports for 1.4: Channels: http://ftp.iq-labs.net/realtime_channels/ Queue callers: http://ftp.iq-labs.net/realtime_queue_callers-1.4/ Queue members: work in progress, needs some refactoring and optimization to make that effective. Meetme: planned, no patches yet To use any of those patches, you will need to add backport of store/destroy to 1.4: http://ftp.iq-labs.net/realtime_store_destroy-1.4/ *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update DB on ringing/ catch ringing event
On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html This should give you up-to-date list of channels in database, so you can use SELECT * FROM channels WHERE state=Ring; to get currently ringing channels. If You find this patch useful, please add a comment to issue http://bugs.digium.com/view.php?id=12556 that you would like to see Realtime status implemented in future versions of Asterisk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] update DB on ringing/ catch ringing event
On Thu, May 8, 2008 at 12:34 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Atis Lezdins schrieb: On Wed, May 7, 2008 at 5:43 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: Benjamin Jacob schrieb: Anyway in Asterisk to update a DB/ do some action on events like ringing. The issue is I need to be able to hangup/cancel a call, if it's ringing(decided by the admin). This is independant of the timeout that we can specify in the Dial command. If I could somehow update a DB with the channel name on ringing, it would solve my problem. I assume NVlinedetect is one way to do it, but that isn't visible anymore, more so for Asterisk 1.4 and above. Any bright ideas on this one? I think there is no other solution but to listen to events on the Asterisk manager interface. For now, not really. You could try Realtime Channels patch I just mentioned here: http://lists.digium.com/pipermail/asterisk-users/2008-May/211136.html Yeah, of course you can do almost anything with a patch. Well, this wasn't specifically written for this requirement. I just want to add some general usage realtime status in Asterisk, and I need user support there :) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 1:07 AM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 07 May 2008 16:11:05 Atis Lezdins wrote: However I encountered a resistance from Asterisk developers, as they don't wish to accept my patches - because they don't wish to support another interface. As I read in Bug Guidelines, if enough users request this, it should make into asterisk, so I'm asking You to express Your opinion on those features. That's not quite correct, either. First of all, the correct forum for this is the -dev list, where we discuss development issues. Second, we gave you an alternative way to do this. You could do this with AMI, with the addition of a single query to access current state, then monitor status continuously for updates. And third, it doesn't make a difference how many users request a particular interface -- the development team has to maintain it afterwards, and if you're proposing a new interface, you need to convince the development team that it's worth the extra hassle -- not the users. True, but resistance I encountered gave me impression that there's no way how to convince devs except lot of users asking for this, so i want to see who would find this useful. I hope that this would convince the development team. So we're not opposed to the concept; we are opposed to the particular interface that you chose to use. Modify it, and it will make its way back into Asterisk. Stubbornly stamping your foot and insisting that you have the right way, and the status quo will remain. Unfortunately the concept I'm offering is that There's no need for continuous AMI connection. Current state can be retrieved already (but that needs locking), and incremental updates are available too (but that needs continuous AMI connection). So all together - I'm saying there could be really simple interface for all this - no troubles with locking of lists or keeping persistent connections. Why would user application need to take care of all this, if DB engine can do that. *** Supporting this feature If You find that those features would be good for merging into Asterisk, please write a comment in bugtracker: http://bugs.digium.com/view.php?id=12556 Please don't. We've already discussed this to enough detail, and if you choose to modify your code, it will show up in the next major release of Asterisk. I understand that code have to match certain standards, coding guidelines and architecture. I'm willing to do any of this, but so far i see all the discussions are about concept. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime status feature - user feedback needed
On Thu, May 8, 2008 at 3:49 AM, Ex Vito [EMAIL PROTECTED] wrote: On Thu, May 8, 2008 at 1:23 AM, Benoit Plessis [EMAIL PROTECTED] wrote: Tilghman Lesher a écrit : Your question leads to this question: why don't you create a proxy application that listens on AMI and populates a database outside of Asterisk, then do all your queries to that database? That would provide exactly the same functionality, but it would not require a single change to the Asterisk codebase. You could even contribute that application back as something in the contrib/scripts subdirectory. True, that was one of initial options, however I prefer to NOT have yet another layer. I will consider this as an option where appropriate. However this looks quite awkward to me, somehow it reminds me tailing queue_log or CDR and putting result into MySQL database.. just one level more that way. For now, I see only one point against this - having status cleared upon module load/unload makes it easier to follow restarts/module loads. I second that, If there is already a way to do things, why adding another one, especialy if it's for caching reasons. While we cannot say that asterisk fall into the KISS rule, it's not a reason to let it grow. Agreed. There should be ONE to do it, it should be SIMPLE and as RELIABLE as possible, without interfereing (bad spelling?) with asterisk's operations: the proxy into AMI looks like the way to acheive the required funcionality... After all, that's exactly the purpose of AMI ! Let's keep the codebase as small as possible, let's make asterisk as solid and reliable as possible. Let's not reinvent wheels! Ok, so we're exactly at the point. Yes, I agree that it would act nearly the same way as AMI actions, however there's one great advantage - It would be really easy to set this up for user. AMI proxy would take more effort, need configuration, etc. Then there should be much more development support for proxy than for code within asterisk (if you have noticed, there's no new code, just reusing existing functionality) I think that there should be several ways how to do something, not just one. Having realtime status won't mean that much changes, for now I can see only 4 families for this - queue_members (already existing), queue_callers, channels and meetme. Really nothing more to give full overview of Asterisk Status. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Queue: tricky problem with MOH
On Thu, May 8, 2008 at 11:25 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
I have this simple queue for the reception set up such that the console
queue has only one agent.
I checked the number in the queue and if there is someone there, I play
back a busy please be patient message and then join the call to the
queue.
If there is no one in the queue, the caller will go directly into the
queue and the receptionist phone will ring. This looks fine but while
the call is waiting for the receptionist to pick it up, the caller will
actually hear Music on Hold instead of just ring ring ring. This is
undesirable.
exten =
7100,n(rcl_off_opn),Set(rcv_que_num=${QUEUE_WAITING_COUNT(console)})
exten = 7100,n,GotoIf($[${rcv_que_num} = 0]?rcl_que_jon:)
exten = 7100,n,Playback(rc-busy)
exten = 7100,n(rcl_que_jon),Queue(console)
exten = 7100,n,Wait(2)
exten = 7100,n,HangUp()
Queue(console,r)
would do what you want, but so you would need to have two entry points to queue.
Regards,
Atis
So, the issue is MOH is good for the 2nd and subsequent callers but not
for the first caller who should just hear ring ring ring until the
receptionist picks up the call.
Any thoughts?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
On Sun, May 11, 2008 at 8:24 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote: GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? GREAT IDEA! (even if it wasn't yours ;-) I have had so many issues with this and desk phones, cell phones being out of range, turned off, or answering machines set to answer after two rings. If this gets implemented, it would be a great feature and save me tons of complaints and explanations. Maybe a posting on the dev list is appropriate. I would certainly contribute to a bounty. Wouldn't a answer macro do exactly what required. It should be executed before bridge, so ANSWER shouldn't be passed upon it's completed. It can read some tone from keypad, and if that confirms, continue by bridging channels. So, this should work with at least queue in ring-all mode (i feel that it would be correct if Dial would do that too) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
On Sun, May 11, 2008 at 8:49 PM, Matt Watson [EMAIL PROTECTED] wrote: I just took a quick look at the dialplan that freepbx uses for doing call confirmation... the dialplan part of it is actually quite simple... its just a matter of setting the USE_CONFIRMATION varialbe =TRUE. However, the actual magic looks like it happenes through its dialparties.agi... which is a little more complicated than i'd like to try and dissect on a sunday afternoon! but that might be a good place to look at how its done to learn by example. It should be like Dial(SIP/123SIP/456,30,M(confirm)); and macro named confirm that playback the prompt, reads DTMF, and sets value of MACRO_RESULT I know in the freepbx implementation what it does is whenever a handset thats part of the ringgroup answers, they get a recorded message You have an incoming call, press 1 to accept maybe it says something else too... can;t recall at the moment. The first member of the Ring group to hit 1 gets the call... if more than 1 person picks up the handset right away, the first to hit 1 gets it, and the rest hear a sorry, too late, somebody else got it-type message (no idea what it actually says). I suppose just a disconnect, because call was already bridged. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No-mobo PC for USB Drives Enclosure?
On Wed, May 14, 2008 at 10:41 AM, Stelios Koroneos [EMAIL PROTECTED] wrote: As people have sugested the ATX power supplies can work without a mobo One thing to watch out for your setup is the actual ampere requirments for your disks i.e Your power supply provides 300W but this is partitioned to different voltages (+5, +12, etc) with different amp charecteristics Disks need 2 voltages. One for the logic (+5V) and one for the motors (+12V) and have different current requirments. Most disk (if not all) mention these ratings on the labels they have What you must do, is to see if by adding the current requirments seperatly for +5V and +12V, does not exceed the power supply's amp rating *for that voltage*, allowing also for a 15% -20% margin, as power consumption will be higher than the nomimal mentioned during disk startup (and you will be starting all your disks at the same time) Also make sure your box has sufficient cooling and there is some short of airflow over the disks, as the number 1 enemy of disks is high temperature and stacking so many disks in a box will create large amounts of heat. I would suggest you to get a good (aka expensive) 500W power supply and use 10-12 disks with it to avoid problems in the long run, Also keep in mind that MTBF specs of SATA disks does not make them an ideal candidate for 24/7/365 operations Another thing is voltage feedback. The Gray wire should be grounded when +5 and +3.3 V is ok for m/b. As +5 is shared also for disk connectors, there could be some problems. Also be advised that you should buy good power supply, as the difference is in voltage stability, and hard disks don't like floating voltages much. I would suggest you to go better for some network oriented setup, use NFS ir CURL for configs, etc. Imagine what will happen if that one PSU will fail. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom XML Files / asterisk
On Thu, May 15, 2008 at 10:08 PM, Robert McNaught [EMAIL PROTECTED] wrote: The way I understood it is that TFTP does not allow you to set a username and password in a URL like tftp://username:[EMAIL PROTECTED] is not possible when setting option 66 Is it not possible to require a username and password with HTTP? I assumed that you could just like if you were protecting the web root directory on a webserver to require authentication credentials, although have never tried this. You can always limit access to HTTP for certain IP range. Isn't that enough? Then add auth in your request string - for example: http://provisioning.mysite.com/secure/234sdfsdf3247sd/- unless you enable directory listing, it should be at same security level as http with authentication or ftp (any of those can be sniffed) Another thing I like in HTTP - you can redirect config read to execute any script, write simple PHP that will generate resulting config, with lookup of correct extension by MAC. Much like DHCP. Regards, Atis Robert On Thu, May 15, 2008 at 10:43 AM, Anthony Francis [EMAIL PROTECTED] wrote: I am confused how TFTP is less secure than HTTP. TFTP does not allow any browsing, ever. Neither technologies will allow the device to authenticate before downloading a configuration file, and both are easily secured by only permitting connections from specific hosts. Robert McNaught wrote: Yes, perhaps a script would always be better than hand-touching these files, and getting an XML editor only really makes it easier on the eyes. On the same subject, I have noticed that Snom and Linksys phones do not support FTP provisioning - only TFTP and HTTP. With TFTP being an insecure option for a hosted architecture, is everyone moving to provision Polycoms with HTTP, so that both can be auto-provisioned via Option 66. One thing I found is that, with option 66 in a LAN router, you cannot specify more than one protocol. Has anyone had any problems provisioning Polycoms with HTTP? On Thu, May 15, 2008 at 1:35 AM, Philipp Kempgen [EMAIL PROTECTED] wrote: Robert McNaught schrieb: Does anyone know how to apply a style sheet to the polycom automatic provisioning XML files? Why should applying a stylesheet be different than for any other XML files? Even better, does anyone know of a web-based XML editor where you can just edit the files from a browser directly ie entering in phone number, display name, proxy address etc. From what I gather, most people are just using Notepad to change the files then upload them, or vi from the command line, which is fiddly and time-consuming. Just use your preferred editor. Nobody forces Notepad or vi upon you. Even better: Generate the config files with Perl/PHP/insert favorite language. Grüße, Philipp Kempgen -- Asterisk-Tag.org 2008, 26.-27. Mai - http://www.asterisk-tag.org amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dear asterisk-users@lists.digium.com May 87% 0FF
On Fri, May 23, 2008 at 3:19 PM, Steve Totaro [EMAIL PROTECTED] wrote: On Fri, May 23, 2008 at 5:49 AM, VIAGRA (R) Official Site asterisk-users@lists.digium.com wrote: About this mailing: You are receiving this e-mail because you subscribed to MSN Featured Offers. Microsoft respects your privacy. If you do not wish to receive this MSN Featured Offers e-mail, please click the Unsubscribe link below. This will not unsubscribe you from e-mail communications from third-party advertisers that may appear in MSN Feature Offers. This shall not constitute an offer by MSN. MSN shall not be responsible or liable for the advertisers' content nor any of the goods or service advertised. Prices and item availability subject to change without notice. (c)2008 Microsoft | Unsubscribe | More Newsletters | Privacy Microsoft Corporation, One Microsoft Way, Redmond, WA 98052 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Is this the new DAHDI Viagra? I think, spamfilter should ban every message mentioning Microsoft :p Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proposed changes for queue timeout
Hello, I've been annoyed quite some time by behavior of queue timeout (specified as argument to Queue app). Basically if I specify timeout for queue 5 minutes, and ring time to agent for 15 seconds, and ring to agent starts at 4:59, agent will receive ring only for 1 second, after which call attempt will terminate. So, the question is - if anybody needs exact queue timing, with possibility that agent calls are terminated without finishing ring timeout? Please see issue http://bugs.digium.com/view.php?id=12690 - there's table of calculations, which explains how values are calculated now, and how I'm proposing. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] detecting which party hung up
On Thu, Jun 5, 2008 at 6:57 PM, Lenz [EMAIL PROTECTED] wrote:
Hello list,
I have a problem that looks quite simple but I cannot find a way to fix.
I have a Dial() command and want to detect which party of the call hung up
- if it was the caller or the callee.
In the dialplan, I have the folllowing commands...
exten =
exten = _9XXX.,n,Dial(${MY_TECH}${MY_NUM}||M(call-answer))
; Trapping call termination here
exten = h,1,NoOp( Call exiting: status ${GLOBAL(${GM})} DS:
${DIALSTATUS} HU: ${HANGUPCAUSE} )
I set the ${GLOBAL(${GM})} variable through a macro 'call-answer', and it
works fine for detecting if the call was answered or not (I have other
logic to run at answer time so it fits me okay).
I thought that there would be a way for me to know on the calling channel
if the 'h' was enetered because this channel hung or because the other
bridged party hung, so I tried ${DIALSTATUS} and ${HANGUPCAUSE}, but they
are always the same no matter who hangs up. Am I missing something here?
Thanks
l.
Hi,
add g flag to Dial app, that way Dial will continue to next priority
when ANSWERED but called party hanged up. However if caller will hang
up, channel will jump to h extension.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Camp / Callback feature in 1.4
On Tue, Jun 10, 2008 at 5:34 PM, Phil Knighton [EMAIL PROTECTED] wrote:
Hello
I'm looking for a way to do the following using my Asterisk system and Snom
SIP phones...
Scenario:
Caller on Internal Phone 1 calls internal phone2. Phone 2 is busy (or more
accurately goes straight to voicemail).
Caller on internal phone 1 can press a button / dial a code (explained in
next step) and hangup
When phone 2 is free, phone 1 rings and on answer dials phone 2
I was sure this was called camping - but all the camping stuff I can find,
refers to the caller having to hang on the phone and wait. Am I missing
something?
Anyone have a solution?
Quick solution that comes into mind:
Set(exten_copy = ${EXTEN});
Dial(SIP/${EXTEN})
if (${DIALSTATUS}=BUSY) {
// prompt for camp
Set(DB(camp/${EXTEN}/call_to)=${CALLERID(num));
}
h = {
Set(call_to=${DB(camp/${exten_copy}/call_to)});
if (${call_to}!=) {
Set(DB(camp/${exten_copy}/call_to)=);
System(call_to ${exten_copy} ${call_to});
}
}
So, in case if phone2 is busy, store callerid of phone1 in database,
so when phone2 will hangup it will triger a script call_to which
however can originate call trough manager or call-file.
Of course you will need some additional handling in case if multiple
callers decide to camp, or diferent protocols are used, etc.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? No, next time it will write new record from the point when ResetCDR was called. I use it extensively for call event logging, for example: * Call received to DID A, business hours detected. * Call sent to IVR 1 for 15 seconds * Call waited in queue 2 for 20 seconds etc Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
On Thu, Jun 12, 2008 at 9:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
Combine GROUP/GROUP_COUNT with category of ${EPOCH}
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group
Calls will still be received by asterisk, however you will be able to
kick them off without proceeding with following dialplan logic.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple CDRs for one call (multiple dial attempts during one call)
On Thu, Jun 12, 2008 at 10:51 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 9:14 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Atis Lezdins wrote: On Thu, Jun 12, 2008 at 3:36 PM, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, I have setup an asterisk system which: recieves incoming sip calls ask the caller the number they want to dial, and then dial that number after the caller is done talking and callee hangsup or even if the callee does not answer the phone, the caller is asked for another number to dial. And so onuntill the caller hangsup Everthing above is working fine. But i dont know how to manipulate the cdr so that every outgoing call for he caller should be logged. I have looked into ForkCDR but it seems like it can only be used for transfers. Any ideas how i can solve my multiple cdr problem? ResetCDR(w) Regards, Atis I'm not sure that would be a viable solution, the ResetCDR(w) app+option is only going to write the cdr and then zero it out, but the next time the write occurs wouldn't it just overwrite the existing record? No, next time it will write new record from the point when ResetCDR was called. I use it extensively for call event logging, for example: * Call received to DID A, business hours detected. * Call sent to IVR 1 for 15 seconds * Call waited in queue 2 for 20 seconds etc Regards, Atis Ah thanks Atis! I hadn't played with it before since the documentation gave info that lead me to believe it wouldn't work for me :) Very helpful information :) You're welcome :) Oh, btw, you will definitely need to enable unanswered = yes in cdr.conf as after ResetCDR new entry has disposition NO ANSWER, even if call is answered before. So without this you could loose them. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk calls per second
Hi,
I already gave a hint into right direction, but seems that it got
missed, so basically it would look like this:
exten=_3XX,1,Set(GROUP()=${EPOCH})
exten=_3XX,2,Set(GROUPCOUNT=${GROUP_COUNT(${EPOCH})})
exten=_3XX,3,GotoIf($[${GROUPCOUNT} ${MAX_CALLS}]?120)
exten=_3XX,4,Dial(SIP/${EXTEN})
exten=_3XX,5,Playback(unavailable)
exten=_3XX.,6,Hangup
exten=_3XX,120,Playback(try-later)
exten=_3XX,121,Hangup
Epoch is UNIX timestamp, which changes every second. Probably you
don't even need to use GROUP, but can keep counter for current second
in some database, however that would need database cleanups and locks.
Asterisk builtin DB wouldn't be useful, as it can't increment within
same operation, so some sort of SQL magic should be used. For example
multiple primary keys, one of which is autoincrement, or just
transactions.
However advantage of using GROUP would be that if call disconnects,
it's not counted within GROUP_COUNT anymore, so you can accept one
more call for that second(probably most useful for minute).
Regards,
Atis
On Fri, Jun 13, 2008 at 3:57 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
Hi Edgar,
Thanks for the reply. This setting is good for 10 simultaneous calls.
What i really need is 10 calls being done per second but no limit on
simultaneous calls.
On Fri, Jun 13, 2008 at 2:43 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
Well, as I said, you can tell Asterisk to accept until 10 SIP calls,
for example, at ANY TIME (I don' t understand why per second, I mean,
if the 10 calls are established in the same second, they are acepted,
and so they are if they are established in the same milisecond, while
the max concurrent calls is belowthe limit of 10).
You can do something like this in your dialplan (assuming extensions like
_3XX)
exten=_3XX,1,Set(GROUP()=sip-calls)
exten=_3XX,2,Set(GROUPCOUNT=${GROUP_COUNT(sip-calls)})
exten=_3XX,3,GotoIf($[${GROUPCOUNT} ${MAX_CALLS}]?120)
exten=_3XX,4,Dial(SIP/${EXTEN})
exten=_3XX,5,Playback(unavailable)
exten=_3XX.,6,Hangup
exten=_3XX,120,Playback(try-later)
exten=_3XX,121,Hangup
where ${MAX_CALLS} is a variable defined by you that is the limit of
calls to be accepted
On Thu, Jun 12, 2008 at 12:16 PM, Mark Quitoriano
[EMAIL PROTECTED] wrote:
yeah something like that. is it possible to set asterisk to make 10
calls per second?
On Thu, Jun 12, 2008 at 8:23 AM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I know you can limit the total calls in any given time, for example,
you say I would like to have 10 SIP calls established as maximum.
On 6/11/08, Mark Quitoriano [EMAIL PROTECTED] wrote:
Is there a way to limit or set the calls per second on SIP?
--
Regards,
Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Mark Quitoriano
Blog | http://mark.quitoriano.org
VicidialNOW! | http://www.vicidialnow.com
APUG! | http://asterisk.org.ph
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Idiot's Question
On Sat, Jun 14, 2008 at 8:46 PM, Venefax [EMAIL PROTECTED] wrote:
Believe it or not, I cannot find online a single piece of documentation for
the Asterisk function SPRINTF. This example does not work, for it changes
the caller id.
Set(CALLERID(num)=${SPRINTF(%010lld,0${CALLERID(num)})}),
For instance, if the incoming caller id is 17864335989, I get 0684466805 out
of that function, which is not intended one. To be precise, of the caller
has less than 10 chars, I want to complete it with a string of '0's. If the
caller id is nothing, or empty, I want to replace it with 10 zeroes. I guess
I can figure it out if a link to the documentation of SPRINTF is provided.
Well, 10 chars or 4294967296 to be precise is the limit of integer, so
on 32 bit platform this won't work. Just do the string processing :)
Btw - some kind of str_pad function in dialplan would be nice ;)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cdr-custom/Master.csv rotation
On Sun, Jun 15, 2008 at 10:23 AM, Tzafrir Cohen
[EMAIL PROTECTED] wrote:
On Sat, Jun 14, 2008 at 08:49:04PM -0700, Darryl Dunkin wrote:
It's like asking for directions, and someone tells you to drive,
useless.
Here is what we do here:
Create /etc/logrotate.d/asterisk:
/var/log/asterisk/asterisk-verbose /var/log/asterisk/messages
/var/log/asterisk/debug /var/log/asterisk/queue_log {
daily
rotate 7
compress
missingok
notifempty
sharedscripts
postrotate
/usr/local/bin/log_rot_ast
endscript
}
/usr/local/bin/log_rot_ast contains:
#!/bin/sh
/usr/sbin/asterisk -rx 'logger reload' /dev/null 21
logger reload rotates logs. But not CSV . That's because the CSV CDR
files are not held open.
If they are not held open, you can can just move them away with mv,
next CDR should just write new file.
Regards
A,tis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Agents getting stuck busy
On Mon, Jun 16, 2008 at 12:30 PM, Kyle Sexton [EMAIL PROTECTED] wrote: Having a weird issue with some agents getting stuck busy on my system. Call will come into the queue and the agent will hit DND, or be DND when the call comes in (DND being the button on eyeBeam softphone, not a star code). After the agent comes back from DND they will be stuck as busy in the queue and I have to reload chan_agent.so in order to get them available. I'm running Asterisk 1.4.17, and the bug sounds a lot like http://bugs.digium.com/view.php?id=9618 but that bug looks to be fixed in 1.4.17. I could suggest you trying on latest version (currently 1.14.21) or at least try this patch http://bugs.digium.com/view.php?id=12127 The description doesn't match your issue, however there was found old code handling dialstatus and translating it to agent state, which could be cause of your problem. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT How Digium Saved My Bacon!
On Tue, Jun 17, 2008 at 10:03 AM, Steve Totaro [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 2:56 AM, Atis Lezdins [EMAIL PROTECTED] wrote: On Tue, Jun 17, 2008 at 6:45 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Florell wrote: Hello, I guess I am one of the lucky few to have one of these handy screwdrivers and it saved me when my son(aged 2) somehow locked himself in a bedroom and couldn't unlock the door. The door knob needed a very small slotted screwdriver to twist-unlock the door and the Digium tweeker(which was also in my pencil cup) saved my bacon as well that night :) Any chance of more of these being handed out at Astricon this year? Thanks, MATT--- On 6/16/08, Mark Hamilton [EMAIL PROTECTED] wrote: Now you're just trying to get us all jealous, Steve. No good. But I'd like that screwdriver! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: June 16, 2008 8:41 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT How Digium Saved My Bacon! I had a laser pointer and power point controller device but the Digium logo rubbed off after a week I do have a t-shirt though Thanks, Steve T On Mon, Jun 16, 2008 at 8:36 PM, Andrew Kohlsmith (lists) [EMAIL PROTECTED] wrote: On June 16, 2008 07:22:18 pm Mark Hamilton wrote: How come he has it, and he's in Paris! I'm in Toronto, and I don't have it? Yeah, me too. I even got a mention in the book, but no screwdriver? :-( -A. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users All I have to say is Murf, SEND ME ONE I'll do anything (within reason) ;-) AEL bug reporting, improvement suggestions, hell I debug and report on the entire new CDR/CEL branch :) ROFLno seriouslyI want one ;-) How about sending those out when certain amount of karma is reached? ;-) Regards, Atis It seems you get these goodies at Astricon events. Unfortuneately it's too far and too expensive for me to get there. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reg call recording
On Tue, Jun 17, 2008 at 8:34 AM, Sherwood McGowan
[EMAIL PROTECTED] wrote:
Bikrish Amatya wrote:
Hi all
I am using asterisk as pbx for my company. My company has requirement
that all the incoming and outgoing calls should be recorded for all the
extensions and should be able to play recorded call on extensions basis,
that is , say 123 extension has made what call on the particular date
and should be able to play and listen to it. What is the better way to
achieve this? Any kind of suggestion is truly appreciated.
Bikrish
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
A simple web interface, such as asterisk-stats coupled with some basic
modifications to link to a recording that was made with ${UNIQUEID} as
the recording filename (pre extension, use monitor + soxmix to mix the
recordings) will work just fine, I use it on a medium-large installation
that does about 10K calls a day, with no issues in regards to recordings
or ability to access calls/recordings.
I have similar setup, and here are some suggestions from my experience.
Do recording only in native format, that will decrease the load by
transcoding at working time.
Whenever somebody requests to listen, you can mix, transcode and play.
This usually takes few seconds (however depends on call duration). Mix
and transcode (to some lower bandwidth codec) the rest of recordings
at night time.
Personally I record everything in ulaw, and either on listen or at
night transcode to gsm for storage.
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI show queues NOT WORKING WELL
On Thu, Jun 19, 2008 at 10:06 PM, Chento Arohuanca [EMAIL PROTECTED] wrote: Just about 30 minutes that I can´t get real information from my Asterisk box. All agents seem to be available but is not true: QUEUE_01 has 0 calls (max 100) in 'rrmemory' strategy (0s holdtime), W:4, C:0, A:0, SL:0.0% within 0s Members: Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 1 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED]/n with penalty 2 (dynamic) (Not in use) has taken no calls yet [EMAIL PROTECTED] asterisk]# asterisk -rx core show channels Channel Location State Application(Data) SIP/641-08cef808 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/641|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/65-1) Zap/65-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/64-1 [EMAIL PROTECTED]: Up (None) SIP/625-09766788 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/625|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/66-1) Zap/66-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_02|tT|||1800) SIP/620-09358088 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:10 Up Dial(SIP/620|120|rtT) Local/[EMAIL PROTECTED] [EMAIL PROTECTED]:1Up Bridged Call(Zap/63-1) Zap/63-1 [EMAIL PROTECTED]:1 Up Queue(QUEUE_01|tT|||1800) Zap/94-1 (None) Up Bridged Call(SIP/623-b2b1d070) SIP/623-b2b1d070 [EMAIL PROTECTED]:3 Up Dial(Zap/g3/2714269||tTrRS) SIP/615-08a892c0 (None) Up Bridged Call(Local/[EMAIL PROTECTED] Please help me with this issue! Local channels don't support state information in Asterisk 1.4. For that you either need to use 1.6 or backport of state_interface for 1.4. Then you have to set call-limit for peers, and specify state_interface device when logging in agents. For more information please search for asterisk queue state, as this has been discussed several times. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On 6/26/08, Grey Man [EMAIL PROTECTED] wrote: On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote: This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! Hi, I just wanted to say that we are working on testing our current functionality. We don't use attended transfers, but would like at some point. So, I'll try to report within next week if something else is broken. Hi murf, From some preliminary testing on the CDRfix4 branch it looks like the CDRs for attended transfers are now correct which is fantastic. For blind transfers the CDR for the first call leg is still incorrect with the duration only being recorded up until the point the transfer occurs. What's wrong with that? This fits perfectly for my needs. Is there a way how to exploit this? Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!
On Thu, Jun 26, 2008 at 10:21 PM, Steve Murphy [EMAIL PROTECTED] wrote: On Wed, 2008-06-25 at 22:50 +0100, Grey Man wrote: On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy [EMAIL PROTECTED] wrote: This is just a note that the fixes in the CDRfix4 and CDRfix6 branches are getting closer to being merged into 1.4, trunk, and 1.6.x. If CDR's are important to you, and you ignore this notice, then you deserve what you get! Hi murf, From some preliminary testing on the CDRfix4 branch it looks like the CDRs for attended transfers are now correct which is fantastic. For blind transfers the CDR for the first call leg is still incorrect with the duration only being recorded up until the point the transfer occurs. I did a blind xfer with my snom360, and got these two cdrs with **TRUNK**: Eventlist: 1. 101 dahdi (used to be zap) phone picked up and 200 is dialed for the snom360 2. 200 (snom360) picks up and answers the call 3. 200 (snom360) hits the Transfer button (101 gets MOH), dials 202 4. 200 (snom360) hits the checkmark button to send off the call (101 starts hearing ringing, 200 starts getting congestion). 5. 202 (eyebeam) answers (101 202 are connected) 6. 101 or 202 hang up. Conversation finished. fxs.01 101,101,200,extension,DAHDI/1-1,SIP/snom360-082c3f68,Dial,SIP/snom360,30,2008-06-26 11:04:08,2008-06-26 11:04:12,2008-06-26 11:05:56,108,104,ANSWERED,DOCUMENTATION,,1214499848.11,, fxs.01 101,101,201,extension,DAHDI/1-1,SIP/murf-eyebeam-082d95d8,Dial,SIP/polycom430SIP/murf-eyebeam,30,2008-06-26 11:06:06,2008-06-26 11:06:12,2008-06-26 11:06:56,50,44,ANSWERED,DOCUMENTATION,,1214499966.13,, Here are the two CDR's with their recorded event times: CDR start answer end 112 3 245 6 above, I called into the snom360, and hit the Transfer button, dialed 201, and got congestion (101 gets moh until I hit the check key), and hung up the snom (200). 201, the eyebeam, rings, I answer. 101 and 201 are connected. 101 hangs up, and the conversation ended. THE SAME PROCEDURE ON THE CDRfix6 branch: fxs.01 101,101,200,extension,DAHDI/1-1,SIP/snom360-0829e2d0,Dial,SIP/snom360,30,Tt,2008-06-26 12:16:37,2008-06-26 12:16:44,2008-06-26 12:17:01,24,17,ANSWERED,DOCUMENTATION,,1214504197.4,, fxs.01 101,101,202,extension,DAHDI/1-1,SIP/murf-eyebeam-082c2b70,Dial,SIP/murf-eyebeam,30,Tt,2008-06-26 12:17:01,2008-06-26 12:17:14,2008-06-26 12:17:49,48,35,ANSWERED,DOCUMENTATION,,1214504197.4,, CDR start answer end 112 4 245 6 Well, time 3 does get lost, but I thought it might be nice to be able to link 1 2 by the coincident times and say, hey, that looks like a blind transfer! One point of dissatisfaction I have with these is the fact that SIP/snom dialed the second CDR, not DAHDI/1. But, if I change it, you won't know that DAHDI/1 was the guy that murf-eyebeam was talking to... tough choices. So, I take it from your above words, that you'd like the 1,2,3; 4,5,6; times on the two CDR's? Can anyone lab this up for 1.2; I don't have enough phones, and I'm not eager to reconfigure the ones I've got for just one test ! I wonder how is this reflected in cdr_addon_mysql. It would show just duration and billsec (at least for 1.4), so i would defineately want this 1 second between 3 and 4 to show up in some record (preferrably in second CDR, as it's not talking time with first user anymore). Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.21 + Realtime Queues = Agents Not Ringing?
On Sun, Jun 29, 2008 at 7:02 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Sherwood McGowan wrote: Gentlemen, I'm using 1.4.21 SVN Tag, and have the queues set up to use Realtime. This system works fine with 1.2.28, and everything loads fine with no errors, but when I log an agent in I see the extra message (not in use) by their listing and they are not rang by asterisk when their queue is called. Any ideas? Nobody else? Have you checked call-limit and state information for SIP peers? That was changed between 1.2 and 1.4, and could affect queue state. See the UPGRADE notes. Otherwise You'll have to set core set debug 2 and core set verbose 3, and post full log (debug+verbose) where agents got logged in (if you have also realtime members, just execute queue show on CLI. Then you'll have to give one call to agent, talk for little and disconnect. Then just post that log here. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QueueMemberStatus
On Wed, Jul 9, 2008 at 12:00 AM, Jason Dixon [EMAIL PROTECTED] wrote: On Tue, Jul 08, 2008 at 11:00:43AM -0400, Jason Dixon wrote: On Tue, Jul 08, 2008 at 12:10:05PM +1200, Matt Riddell wrote: Action: Command Command: show queue my_queue_name ActionID: my_queue_name_12345 This does not appear to show the correct status of an extension. It appears that ExtensionState also always reports Status of -1. Are there any Actions or Commands that will report the correct status of an extension? So far the only accurate representation I've found of queue members has been the following. $ sudo /usr/sbin/asterisk -r -x show channels | grep '^SIP' SIP/241-b742e010 [EMAIL PROTECTED]:2Ring Dial(Zap/G1/411) $ sudo /usr/sbin/asterisk -r -x show queue support_queue | grep SIP SIP/207 (Ringing) has taken no calls yet SIP/203 (Not in use) has taken no calls yet SIP/202 (In use) has taken no calls yet SIP/201 (Not in use) has taken no calls yet All of the commands I've tried via the AGI have yielded incorrect results. If this sounds wrong, please let me know and I'll resume beating my head against the nearest wall. :) There is QUEUE_MEMBER_COUNT (in 1.4) and QUEUE_MEMBER (in 1.6) dialplan functions which allows to get count of members (in 1.6 also count of free / logged in members). You can use GetVar to evaluate that. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] finding out on hold channels
On Fri, Jul 25, 2008 at 2:59 AM, Al lists [EMAIL PROTECTED] wrote: I noticed that i' m not getting any manager event for hold and unhold of a channel. is this normal? Also is there any easy way through either CLI or manager to find out which one of the channels are on hold? I checked show channels that did not show a channel being on hold or not, also sip show channels does show that but it has call id instead of channel id. Hi, There was recently a thread regarding this on asterisk-dev (http://lists.digium.com/pipermail/asterisk-dev/2008-June/033466.html). There was message explaining how to do this by adding custom code to Asterisk sources, and I guess it could be already done in trunk. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Customized Queuing Strategy
Sorry for previous blank answer :)
On Mon, Aug 4, 2008 at 1:20 PM, Syed Nasruddin [EMAIL PROTECTED] wrote:
Hi Thanks ALL for reply,
If I use cascading queue will it do the trick?? The only problem is (as
mentioned in below example) if a call enters testq and get answered then
after hungup at the agent end only will the call will again enter the
next queue which is testq2 as in this example.??
Check the QUEUESTATUS variable: http://www.voip-info.org/wiki-Asterisk+cmd+Queue
Moreover if I keep penalty 1 for all the first 5 agents and penalty 2 or
higher for all the next 5 agents and implement ringall strategy will it
do the same effect??
Yes
exten = 1589,1,Answer
exten = 1589,2,Ringing
exten = 1589,3,Wait(2)
exten = 1589,4,Queue(testq|t|||45)
if (${QUEUESTATUS=) Hangup();
exten = 1589,5,Queue(testq2|t|||45)
exten = 1589,6,Hangup
Regards,
Atis
thanks in advance.
Syed nasr
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Balashov
Sent: Monday, August 04, 2008 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Customized Queuing Strategy
Syed Nasruddin wrote:
1. 10 Call Center Agents.
2. All the calls coming in will ALWAYS be routed to specific 5
agents,
firstly.
4. IF ALL the first 5 agents are busy then ONLY then the call will be
routed to next 5 Agents.
Set up two queues. Call Queue() on the first queue - corresponding to
#1 - with a rather strict timeout. Fall back on the second queue.
More sophisticated strategies require either the modification of the
source code for app_queue, or custom queue implementation in AGI.
--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] skype and Asterisk opensource integration
On Mon, Aug 4, 2008 at 10:58 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all except of some commercial hardware / software gateways, is there any opensource or free project to setup a Skype Account on Asterisk? The only one known to the moment is chan_celliax, which is originally for connecting to cell phones by cable, however it supports also skype (just 1 account). It will launch fake X server and original skype, and communicate with it. http://www.celliax.org/ Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
