Re: [asterisk-users] ACD functionality , Skills for agents
On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote: I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how would I do this? Beware of queue weights. They have caused major problems in the past for many people on this list. As I understand it, enabling weights requires * to grab a lock on a large number of data structures related to queue state, which can cause performance slowdowns and crashes. I haven't seen reports of this recently, so it might be better in the later 1.4 releases, but at one time it was a sure-fire recipe for pain. Also can someone point me to resources for making a single queue with customer calls tagged with agent skills? What I mean is instead of having multiple queues Sales,Tech support, etc, have only a single queue with calls being tagged according to the customer's choice from IVR, so if a customer would choose SALES , the call would go into the queue with other calls but it would only be answered from agents with the skill SALES. This is something offered in other PBX systems like Avaya but im pretty sure it can be done on Asterisk, right? It probably could be, but it would make reporting pretty difficult, as the key fields in the queue log are the call id and the queue name. While you could use the QueueLog() application to stick extra data about the call (e.g the skill chosen from the IVR) into the queue log, that would appear in one line only and require post-processing to glue it together with the rest of the data for that call. I'm pretty sure it wouldn't mesh nicely with the reporting package I use (QueueMetrics). What I do for this is maintain queue (skill) membership in a database, then add the channels to the appropriate queues when the agents log on via a web page. Is there a particular reason you want to just have one queue? -- j. ___ --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] ACD Queue LOG RINGNOANSWER Content 0
On 11/12/07, asterisk [EMAIL PROTECTED] wrote: In my queue log I see that on the RINGNOANSWER Event I get different content. Some events soe the ring timeout (15000). Other events show 0. Other yet show 1000 Doens anyone know what 0 means? Did it try to ring the phone, but it was busy? For my internal reporting, I consider 0 or 1000 to be the result of a phone being on DND. Since all the time values in the log are rounded to the nearest 1000, I speculate that 0 is a rejection in 500ms and 1000 is a rejection in the 500 to 1499 ms range. I figure unless someone is hovering over the ignore button on a softphone, they aren't going to be able to click it so fast that Asterisk registers it as RINGNOANSWER|1000. Likewise, RINGNOANSWER|2 is (for me, given timeout=20 in queues.conf) a failure to pick up a presented call. Everything from 2000 through 19000 I treat as a manual ignore triggered by the agent. So far, the reports I generate based on these rules seem to make sense to the managers reading them. -- j. ___ --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] 'a' extension
On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. You're sending them into VoiceMail() from your dialplan - just stick the dialed number in a channel var before calling VoiceMail(), then refer to it in your 'a' extension. -- j. ___ --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] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
On 11/6/07, Stephen Bosch [EMAIL PROTECTED] wrote: It survives if it goes to a Telus customer, but not if it crosses over to Bell, Rogers, etc. Well -- here's where you can help me, because our name info is not even surviving on Telus' own network. I don't really care too much about Bell and Rogers, since Bell barely has a footprint out here and Rogers doesn't provide CNAM on its mobile network anyway (and nobody is using Rogers home phone ;) ). So -- if you had it working on Telus, what did you do? As soon as I turned on facilityenable=yes, outbound name display started to work for me. One tech claimed it was because I was sending calling name in addition to the IE, He probably meant the Display IE *and* the Facility IE. If you see my post it's what the technician I was working with suggested. Would be great if I knew a way of turning off the Display IE, if that's even possible/allowed. If it's not, then the don't send both idea is wrong. Probably correct, as that's what I'm sending now: [1c 15 9f 00 00 00 00 00 00 00 00 00 00 00 00 00 43 6f 6d 77 61 76 65] Facility (len=23, codeset=0) [ 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 'Namehere'] [00 00 00 00 00 00 00 00 00 00 00] Display (len= 8) Charset: 31 [ Namehere ] I just checked dialing out of and back into my system, and the Facility IE comes back in the SETUP message, but the Display IE does not. Did you end up adding name records to the LIDB? For our purposes name display was a nice to have, so we didn't go down this path. Anyway -- again -- what did you do to get it working on Telus' network? Do you know what kind of switch you were connected to? To the best of my knowledge, we're connected to a 5ESS running NI-2. Here's the relevant zapata bits I use: facilityenable = yes pridialplan=unknown priindication=outofband overlapdial=no resetinterval=86400 echocancel=yes switchtype=national signalling=pri_cpe callerid=asreceived Hope that helps. -- j. ___ --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] Telus (Alberta) PRI Caller ID NAME, Display IE, Facility ID
On 11/6/07, Stephen Bosch [EMAIL PROTECTED] wrote: We are trying to send caller ID NAME information over a Telus PRI in Alberta. The PRI tech says that he sees the NAME information, and for calls over the same network, that NAME info should be reaching the receiving station, but it is not. I've had no end of trouble getting CNAM out of NI-2 PRIs with Telus. We're in Ontario, but the switch configs are the same across the country I believe. It survives if it goes to a Telus customer, but not if it crosses over to Bell, Rogers, etc. One tech claimed it was because I was sending calling name in addition to the IE, while another claimed it was just a problem when the call passes from a NI-2 circuit to NI-1 (which some of the other carriers still use). So, no real solution for you, but at least you know it's not something obvious you're doing. I've tweaked my zaptel settings back and forth and tested with Telus on the phone to no avail. In the end, we deemed the effort to not be worth it. -- j. ___ --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] Dynamic Queue Members - Auto Logoff
On 11/5/07, Nick Brown [EMAIL PROTECTED] wrote:
Another quick question (Spending the day trying to get this project sorted
and tucked away) If I am dynamically adding queue members, they will not
abide to settings within agents.conf will they?
correct.
Ie. I need the equivalent of Autologoff however want my agents to receive
calls when someone joins the queue, not have to sit on hold all day. I see
AgentCallbackLogin has finally been removed.
Has anyone got a work around for this?
It hasn't been removed (in 1.4), just deprecated (I assume you're not trying
this with -trunk). Still, it's not compatible with adding members via
AddQueueMember().
There is an example of doing auto-logoff in docs/queues-
with-callbackmembers.txt in the source distribution. Look for macro
callagent for the specific block that does the work. You do have to be
using Local channels to make this work though, as you need to Dial() the
actual device from the dialplan, then check ${DIALSTATUS} to make decisions
about what to do if the agent doesn't pick up.
j.
___
--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] use dial plan passed arg value in C agi code
On 11/2/07, Arpit Mehta [EMAIL PROTECTED] wrote: Hello * users, I know that passing variable in the AGI script is by exten = _.,1,AGI(simple_c_prgm|123|789) ; 123, 789 are arguments being passed and simple_c_prgm is C code Now how will I receive these variables within C code ? Is it by the same way arguments are passed in command line to C by using argc and argv or there is more to be done than that? From the wiki (http://www.voip-info.org/wiki-Asterisk+AGI): When Asterisk starts an AGI script, it feeds the channel variables to the script on standard input. The variable names are prefixed with agi_ and are separated from their values by a colon and a space. Though the actual channel variables may be in the upper case, the names passed to an AGI script are all lower case. Also, some channel variable names as passed to AGI script differ from the current variable names used in the dial plan. These docs, while dated, are also useful: http://www.bitflipper.ca/Documentation/agi.html It even includes code for getting at variables from several languages. -- j. ___ --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] use dial plan passed arg value in C agi code
On 11/2/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Sadly those docs cover the situation of this:
exten = 666,1,Set(MY_VAR=fred)
exten = 666,n,AGI(simple_c_prgm)
Not this:
exten = 666,1,AGI(simple_c_prgm|123|789)
Looking back over it, you're right.
However, multiple args passed to the AGI work fine; they come in the same
way as any standard argc/argv program.
i.e.:
AGI(agentpause.agi
,--mode=pause,--autopause,--agent=${AGENTCODE},--pausereason=${PAUSEREASON})
I handle this (in Perl) using Getopt::Long, which knows nothing of AGI's
stdin/stdout mechanics:
GetOptions(
\%opts,
'mode=s',
'agent=i',
'pausereason=i',
'autopause!',
) or die can't parse command line;
___
--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] PRI span configuration - span remains down
On 10/26/07, Matthew Fredrickson [EMAIL PROTECTED] wrote: Is there some part of the debug output I need to tell the telco about? When I was on to them earlier today, the engineer only seemed to know how to turn bits of their network on and off, not much about settings I need my end etc. Just tell them when you try to make a call, you get cause code 44 back (channel unavailable). They can look at their switch to figure out what's going on. I had a strange problem with cause code 44 on just 5 B channels of a PRI. The first time I'd dial, I'd get cause code 44 and * would attempt to restart the B channel. The switch would never respond to the request to restart, so the channel remained in limbo from *'s perspective, and further attempts to dial out explicitly on that channel would give me congestion (generated from *, not from the Telco), and attempts to dial out using a group that contained those channels would just skip over them. I called the Telco, and spent over a week trying to convince them that the RELEASE COMPLETE was coming from their end. They claimed it was coming from me. It was almost as if something in between my system and where the tech was running his trace was proxying the Q.931 messages, and sending us both a cause code 44 when I used those channels. In the end, they re-built my trunk and the problems immediately cleared, so it was apparantly some buggered state in their switch. This was with a 5ESS running NI-2 if that helps. -- j. -- j. ___ --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] Voicemail Options
On 10/26/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: I know that you can set it up to where a user hits 0 from their mailbox and goes to an operator, but can you set up other options as well? Could I have 0 for an operator and 1 to go to another extension? I know you can do this by building an AA, but I don't want to have to do that for every user as there are about 40 people that want this. They won't all go to the same number. Thanks. You can also exit VoiceMail() using *, which jumps you to the 'a' extension in the calling context. As for building an IVR for 40 users, you could store the destination in ASTdb or realtime keyed by original extension. Then look up where to send them when they press * based upon the mailbox that VoiceMail() was called against. -- j. ___ --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] Cisco 79xx logon/logoff
On 10/25/07, Adrian Marsh [EMAIL PROTECTED] wrote:
I'd like to know if anyone has figured out a way to be able to have
users logon/logoff manually from Cisco 79xx phones (with SIP firmware
loaded)?
Scenario is, user walks into office, sits at a random desk, and logs
onto the phone. The system would need to log them off of the last
hardphone they were on, and then configure the new phone for their
extension.
We're creating hotdesks and it would be good if users could logon/logoff
the desk phones.
At present they all use softphones on the PC too, and I could engineer a
way of maybe doing things via cgi scripting, replacing the tftp config
files for that phone, and then remotely resetting the phone, however
that would be quite clumber sum.
And before I go that route, I wondered if any of the commercial A*k
systems already offer this?
I haven't done this with Cisco specifically, but I have done it with other
hardphones. I didn't go the route of updating the phone's config while the
person was at that desk because I didn't have a reliable way to remotely
restart the phone.
Instead, I gave each phone an extension and dynamically created links in
ASTdb between the person's extension to the phone's extension for the
duration of the login session. All the hot desk phones are in a context
that when nobody is linked to that desk only allows you to logon and do
basic things like call the operator and emergency services. If a user is
linked to the desk, then I do a
Goto(proper-context-for-that-user,${EXTEN},1), which gives me dynamic
contexts for outbound calling without having to have my sip users in
realtime.
The downside to this approach is MWI, but all of my users get voicemail via
email with delete-on-send enabled, so I just kind of sidestepped the issue.
I'm not sure if you're specifically asking about using the softkeys to do
this without going through some kind of IVR application; worst case you have
speed-dials to your logon/logoff extensions. You could even save a key by
having a single logon/logoff extension that changed its behaviour based on
whether a link exists for that desk. Then again, if someone forgets to log
off it might be easier for users to have a dedicated 'logoff' button rather
than having to press the dual-purposed key twice.
Hope that gives you some ideas.
--
j.
___
--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 tune Asterisk AMD - Answering Machine Detection hacks
On 10/24/07, Costa Dinoteli [EMAIL PROTECTED] wrote:
Most everytime Asterisk calls it thinks it is an Answering Machine and it
starts playing
the AMD message, instead of the delivering the 1st real message
Why is it thinking that it's a machine? If you're on the console at verbose
3 or higher, you'll see what thresholds were tripped. You can also get the
reason in the ${AMDCAUSE} variable:
[Oct 23 09:58:34] VERBOSE[25147] logger.c: -- AMD: Zap/81-1 416XXX
(null) (Fmt: 4)
[Oct 23 09:58:34] VERBOSE[25147] logger.c: -- AMD: initialSilence [2500]
greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000]
minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4]
silenceThreshold [256]
[Oct 23 09:58:37] VERBOSE[25147] logger.c: -- AMD: ANSWERING MACHINE:
silenceDuration:2500 initialSilence:2500
or
[Oct 23 09:43:37] VERBOSE[24313] logger.c: -- AMD: Zap/4-1 4166XXX
(null) (Fmt: 4)
[Oct 23 09:43:37] VERBOSE[24313] logger.c: -- AMD: initialSilence [2500]
greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000]
minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [4]
silenceThreshold [256]
[Oct 23 09:43:38] VERBOSE[24313] logger.c: -- AMD: Word detected.
iWordsCount:1
[Oct 23 09:43:38] VERBOSE[24313] logger.c: -- AMD: Changed state to
STATE_IN_SILENCE
[Oct 23 09:43:39] VERBOSE[24313] logger.c: -- AMD: HUMAN:
silenceDuration:800 afterGreetingSilence:800
Figure out why AMD thinks it's a machine and you can change the thresholds,
either in amd.conf or in the call to AMD().
--
j.
___
--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] Control space of each voicemail box
On 10/15/07, Pepo [EMAIL PROTECTED] wrote: I am using Asterisk like voicemail of a great system with many users, How do I can get statistics of each box in the voicemail system? something like space, number of messages, etc. The only CLI commands are 'voicemail show zones' and 'voicemail show users'; the latter shows the number of new messages for each box, but not the total. So as the previous poster described, shell tools are your friend. If you're storing voicemail using IMAP, then this becomes either easier or harder (depending on your experience with email). You can get more details about the contents of a mailbox using the IMAP protocol, but depending on the implementation you might get less info about the actual bytes taken up in backend storage. -- j. ___ --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 PSTN pickup
On 10/12/07, Yair Hakak [EMAIL PROTECTED] wrote: you'll have to excuse the ignorance (i'm a software guy, not a telcom guy..) Is there any way to know if a channel has been answered by an automatic system (like voicemail) rather than a human being? Specifically, I want to use a .call to make a call on a channel and only do something if a person answers, not a machine of any kind. Is this even possible, or is an answered channel an answered channel? Asterisk 1.4 has the AMD() application that uses several thresholds (which you can vary) to determine if the other end is a human or machine: -= Info about application 'AMD' =- [Synopsis] Attempts to detect answering machines [Description] AMD([initialSilence][|greeting][|afterGreetingSilence][|totalAnalysisTime] [|minimumWordLength][|betweenWordsSilence][|maximumNumberOfWords] [|silenceThreshold]) This application attempts to detect answering machines at the beginning of outbound calls. Simply call this application after the call has been answered (outbound only, of course). When loaded, AMD reads amd.conf and uses the parameters specified as default values. Those default values get overwritten when calling AMD with parameters. - 'initialSilence' is the maximum silence duration before the greeting. If exceeded then MACHINE. - 'greeting' is the maximum length of a greeting. If exceeded then MACHINE. - 'afterGreetingSilence' is the silence after detecting a greeting. If exceeded then HUMAN. - 'totalAnalysisTime' is the maximum time allowed for the algorithm to decide on a HUMAN or MACHINE. - 'minimumWordLength'is the minimum duration of Voice to considered as a word. - 'betweenWordsSilence' is the minimum duration of silence after a word to consider the audio that follows as a new word. - 'maximumNumberOfWords'is the maximum number of words in the greeting. If exceeded then MACHINE. - 'silenceThreshold' is the silence threshold. This application sets the following channel variable upon completion: AMDSTATUS - This is the status of the answering machine detection. Possible values are: MACHINE | HUMAN | NOTSURE | HANGUP AMDCAUSE - Indicates the cause that led to the conclusion. Possible values are: TOOLONG-%d total_time INITIALSILENCE-%d silenceDuration-%d initialSilence HUMAN-%d silenceDuration-%d afterGreetingSilence MAXWORDS-%d wordsCount-%d maximumNumberOfWords LONGGREETING-%d voiceDuration-%d greeting -- j. ___ --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] Mask Initial Processing with Ring Back Tone
On 10/11/07, James FitzGibbon [EMAIL PROTECTED] wrote: What you do in between is up to you. Many people use something like Wait(2) to give a comfort ring, since PRI-connected incoming calls can often be set up nearly instantaneously. You'd want to limit the time obviously, and have proper exception handling in case whatever you're doing between Ringing() and Answer() fails. I should add that some applications (usually ones dealing with audio) will answer the channel for you, so you do have to be cognizant of that. If what you're doing in between is information processing, you should be OK, but I believe calling an AGI will auto-answer the channel, limiting what you can do somewhat. -- j. ___ --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] Mask Initial Processing with Ring Back Tone
On 10/11/07, Victor [EMAIL PROTECTED] wrote: I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? You start sending ringback with Ringing() You answer with Answer() What you do in between is up to you. Many people use something like Wait(2) to give a comfort ring, since PRI-connected incoming calls can often be set up nearly instantaneously. You'd want to limit the time obviously, and have proper exception handling in case whatever you're doing between Ringing() and Answer() fails. -- j. ___ --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] Setting caller id value on outgoing calls using .call files
On 10/4/07, Arpit Mehta [EMAIL PROTECTED] wrote: I was looking at a way to add the caller id to the outgoing calls (which are made using .call files) using asterisk. Any ideas how to do this ? Currently I get 'Unknown' number displayed on my phone when asterisk makes an outgoing call. Add a CallerID: whatever line to your callfile. http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out has a reference of the callfile contents. -- j. ___ --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] show queue (queue name)
On 9/25/07, Everton Goularth [EMAIL PROTECTED] wrote: does anybody know any way that when it run reload app_queue in the asterisk cli it don't lose the informations from show queue (queue name) ? A 'keepstats' option has been added to -trunk, and will show up when 1.6 is released. Until then, you'd have to look at backporting this change: http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=43945r2=44150 (it's a pretty small change) -- j. ___ 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] Hints / State change on outgoing calls
On 9/19/07, Alex Epshteyn [EMAIL PROTECTED] wrote: Also, Asterisk restart results in all the watchers being lost. Is there a way to force the phone to subscribe to notifications after restart (short of rebooting it) and is it phone specific? Usually resubscribe-interval for extensions is client controlled, much like SIP re-register interval. Just make sure it's in between the min and max registration times as displayed in the output of 'sip show settings', otherwise you can run into problems where the phone thinks that the subscription is valid for longer than Asterisk does. -- j. ___ 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] Building an RPM from Asterisk 1.4
On 9/20/07, Marcus Franke [EMAIL PROTECTED] wrote: Do you have any examples for these spec files? I found a repository for installing Asterisk on Centos, but it took a while before I discovered it. Ok, just checked the link its for RHEL, but as Centos is just recompiled this won't matter. Same situation for Ubuntu using the debian package format, but I have not found a repository so far and Ubuntu delivers just the old 1.2 release. :) www.atrpms.net has pretty solid RPMs, and you can grab the SRPMS in order to get the spec file (either install the SRPM or use rpm2cpio to convert the package and extract the specfile manually). This is where I get my libpri and zaptel RPMs from (though I still build * from source, as the RPM compilation options they use are not to my liking). There is a book called Maximum RPM. The dead tree version is now pretty out of date with respect to the latest version of RPM (though still a good introduction if you've never built a package). I believe there was a slightly more up-to-date online version, but it still had some gaps the last time I looked. The best way to learn seems to be to examine good examples and then build your own package using their techniques. -- j. ___ 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] Softphone RTP Session Start-up Delay
On 9/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: here because we are actually specifying the IP Address of the Asterisk server, but I am willing to try anything to fix this problem. The two user pc's are setup on workgroups, so I do not believe that there is a domain available that can be entered in the hosts file. Could the DNS still be the issue? If not, would anyone be able to suggest any other possible problems that may be causing this delay. It wasn't the same magnitude (more like 4 seconds for me), but I had an issue where the default eyeBeam (the commercial version of X-Lite) install was imposing a delay when a call first came in while it attempted to contact a non-existent STUN server. When I removed the STUN server setting, call setup was immediate. Might be worth looking at. Have you done a trace on the PC where the softphone is running (without a filter) to see what network packets are flying at the time the call setup happens? -- j. ___ 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] Why does everyone seem to dislike *now?
On 9/17/07, Jim Canfield [EMAIL PROTECTED] wrote: stuff useless. My real concern was the immediate '/ignore' for asking about an issue with the *now ditro that actually had nothing to do with the GUI itself. Truth be told, most of my time today was in the CLI You may be taking what happened on IRC a bit too literally. You got plenty of help on your Polycom and call parking issues on 09/14, from some very knowledgeable people. On 09/17, you asked one question about rPath/Conary and one person did a '/me puts jcanfield on ignore' emote. He probably didn't even ignore you, it was just his way of saying he wasn't interested in answering questions about packaging systems on a Linux distro he doesn't use. The simple answer to your question (how do I get the LDAPGet module) is answered on the Wiki - you download it from the author's site. The question of how do I package some arbitrary source code into a conary package? isn't really germane to #asterisk. As to the second class citizen point, I think you'll find that people who come into #asterisk asking about problems with their GUI-enabled Asterisk install fall into one of two categories: those who are willing to reduce the problem down to it's non-GUI elements and pastebin the configs and output, and those who are incapable or unwilling to do so. The former tend to get help from people on #asterisk; IMHO the latter should find other places to ask for help, or pay for consulting services. -- j. ___ 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] Enabling MySQL UNIQUE from cdr.conf
On 9/17/07, Luís Palma [EMAIL PROTECTED] wrote: Is there a way to enable the usage of UNIQUEID CDR field using a MySQL database backend for storing CDRs without having to recompile asterisk-addons as stated here http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? After version 1.4 it is said in release that it can be done (not sure if it applies to mysql backend) In addons v1.4.2, it's not possible without recompilation. You get one of two versions of code depending on the definition of a compile time constant. If that constant isn't defined, the text of the SQL INSERT statement in the shared module will be: INSERT INTO %s (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,userfield) instead of INSERT INTO %s (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid,userfield) In the -trunk version of cdr_addon_mysql.c, the behaviour of loguniqueid was changed from a compile-time to runtime option, just like userfield already was. The changes to make loguniqueid a runtime option are pretty small, and trivial to backport to the 1.4 branch on their own. You'd have to do more research to see if you can just build the trunk version against 1.4, given that trunk also has added MySQL SSL support. Of course, if your question stems from the fact that you are unable to recompile anything in your installation, none of this is much help. :( -- j. ___ 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, David Gomillion [EMAIL PROTECTED] wrote: I've stayed out of this thread for a long time, and really didn't read the past comments, so if I'm repeating someone, I'm sorry. I've been thinking this for a while, and just have to say it. If you feel like you have to keep people from turning off the auto-answer feature on a softphone, you don't need a new softphone. You need new people. Yes, but have you ever drawn up a budget for a full-blown meatware(tm) upgrade? Makes Vista look like a picnic. -- j. ___ 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] Agent Callback Login in 1.4
On 9/13/07, Kevin P. Fleming [EMAIL PROTECTED] wrote: It shouldn't be that hard to translate the AEL example into traditional dialplan language; in fact, Asterisk does that itself when you load the AEL into memory, so if you load it yourself and then do a 'dialplan show' you'll see the translated version, which you can then copy into your database. You can also use 'aelparse -w' to dump extensions.ael as extensions.ael.dumpto assist in this. The branching and labeling of priorities is designed for efficiency, not readability, so you'll have to go over it carefully to get a good feel for how AEL constructs are turned into extensions. According to Murf, one of the purposes of this switch was to allow people to write dialplan in AEL and insert it into * installations where AEL was either not supported (1.2) or not viable (GUIs, realtime, resistance to change, etc.). -- j. ___ 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] Force a new user to configure Comedian mail?
On 9/14/07, Jeremy Wadhams [EMAIL PROTECTED] wrote: In Asterisk 1.4, is there any way to force new users to configure their mailbox? I'm thinking a simple IVR that holds a user's hand through changing their PIN, recording their name, and setting up one or both greetings, the very first time they use the account. If your pin is equal to your mailbox, VoiceMailMain() does this automatically when you log in. -- j. ___ 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] Voicemail in 1.4?
On 9/13/07, Ken D'Ambrosio [EMAIL PROTECTED] wrote: I got dragged away from Asterisk (somebody made me an offer I couldn't refuse for system administration), but I'm thinking about seeing if I can't get it deployed at my new employer. Regardless, there are two things about older voicemail that used to annoy me: - Dial by name. Has anyone made it so it can be first or last? Yes, Directory() has a switch to make it search by first name. You still need to choose one or the other, or use the dialplan to ask the user whether they want to search by first or last before calling Directory with or without the switch, but it works. - Jump to voicemail; you used to have to actually dial the voicemail, whereas most voicemail systems allow you to go to your mailbox when you hear your voice prompt. Any chance this has been rectified? Look at the 'a' extension, which will be executed if you hit '*' while listening to the outgoing message. -- j. ___ 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] exit ChanSpy with DTMF
On 9/11/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Part of a supervisor menu I'm writing requires that I allow the supervisor to choose to ChanSpy a channel from the main menu then return back to the menu (dialplan) to choose other options when she's done. Is there a way to 'exit' ChanSpy and continue down the dialplan? Or is a caller stuck in ChanSpy until they hangup the phone? In 1.4, they are stuck. -trunk has an option to allow them to escape out to a context using a DTMF digit; check the changelog in SVN for details. I'm not sure how portable it might be back to 1.4/1.2 if you want to attempt that. -- j. ___ 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] Which cause less CPU usage: GSM or wav??
On 9/9/07, Barton Fisher [EMAIL PROTECTED] wrote: Thanks, OK, a bit confused The cards are TE410P. I really don't see how the set a codec for this, other than it might default to something in code like ulaw. Any clue on how to verify codec in use during a call? If you absolutely want to be sure, use 'pri intense debug span X' and watch for SETUP messages: Protocol Discriminator: Q.931 (8) len=62 Call Ref: len= 2 (reference 542/0x21E) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 95] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 21 ] You'll see the voice characteristics in the Bearer Capability details (I have NI-2, this might be different for NI-2 or other PRI variants). But as others have mentioned, generally T1 PRI = uLaw, E1 PRI = aLaw. -- j. ___ 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] Cascading queues calls not joining unavailable queues.
On 9/7/07, Mark Michelson [EMAIL PROTECTED] wrote: After talking in #asterisk-dev with those who wrote the joinempty option, it appears I was mistaken about its use. joinempty=strict will only not allow a caller to join the queue if the members are in a permanently unavailable state. A member being busy is not enough to trigger the joinempty=strict option. Since DND on most phones works the same way as a phone being busy, DND will not trigger the joinempty=strict option either. It's also important to distinguish between DND-style busy (where in the case of SIP the INVITE is rejected with a 480 temporarily unavailable) and the internal channel state AST_CONTROL_BUSY. For SIP, the latter occurs when the use count exceeds the configured call-limit, and results in no INVITE being sent. The two also have drastically different behaviours when you have members with penalties. Assume we have five agents with penalty zero (A-E) and five agents with penalty 1 (F-J). Assume also that 'ringinuse=no' is set for the queue. If A-D are in use and E is available but their phone is set to DND, then app_queue will continually attempt to dequeue to agent E, never trying agents F-J. If E is busy because they have a call-limit of 2 and they have two calls active, then app_queue will attempt to dequeue to agents F-J. This has been a source of confusion for my users on several occasions. IMO, It also puts an unrealistic burden on agents to always put themselves on pause before they walk away from their desk, and since agents are human and sometimes forget to do this, prevents me from using agent penalties extensively. I just can't explain to non-technical people why one phone keeps ringing while five other agents sit idle. It makes no sense to them. Someone developed a new strategy for app_queue they called XRRMEMORY that will seek to higher-penalty agents after trying all lower-penalty agents. They posted details about it on voip-info: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue (look at the second comment) Unfortunately, the patches weren't done against trunk or the head of 1.4, and the author didn't file a disclaimer with Mantis, so the bug ( http://bugs.digium.com/view.php?id=9165) was recently closed. -- j. ___ 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] Cascading queues calls not joining unavailable queues.
On 9/7/07, Atis [EMAIL PROTECTED] wrote:
Well, for that case i have a RemoveQueueMember() after Dial, in case
of ${DIALSTATUS}!=ANSWERED. That works great, except for agent
complaints - that they are logged out from queue :D Would be a bit
better to be able to set agent's status to Unavailable.
This only works if you're using Local channels to bridge calls to agents.
It doesn't work if you're using AddQueueMember with SIP channels, because
the Dial() is implicit, so you have no control over what happens after that
implicit Dial() finishes.
And yes, I have good reason for using SIP channels. We have externally
driven automatic pausing (because the built-in wrapuptime is per-queue and
therefore broken for any agent who is assigned to more than one queue), and
neither form of Local (with or without /n) perform properly under this
configuration.
It would be great if you could define arbitrary states in queues.conf, then
have dialplan logic to set an agent to given state. If you could indicate
in the definition what things a state means (able to take a call, counted as
part of ${QUEUE_MEMBER_COUNT()}, etc. that would truly be useful.
--
j.
___
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] Cascading queues calls not joining unavailable queues.
On 9/7/07, Atis [EMAIL PROTECTED] wrote: It doesn't work if you're using AddQueueMember with SIP channels, because the Dial() is implicit, so you have no control over what happens after that implicit Dial() finishes. Nop, it works for Dial to SIP channels, if you set g option in Dial. I call AddQueueMember like this: AddQueueMember(queuename,SIP/1234) When I then call Queue(queuename) from my dialplan, the Dial() of SIP/1234 is implicit. There is no priority and extension that I control where Dial() is being executed, and thus there is no priority + 1 for me to go to by using the 'g' option to Dial. What you're talking about only works if I do this instead: AddQueueMember(queuename,Local/[EMAIL PROTECTED]) And then somewhere in my dialplan I have: [agents] exten = 1234,1,Dial(SIP/1234,20) exten = 1234,n,DoSomethingHereIfDollarDialStatusIsNotANSWERED And then of course it makes no difference that agent 1234 is on SIP - they could be Zap, IAX, Skinny, whatever. And yes, I have good reason for using SIP channels. We have externally driven automatic pausing (because the built-in wrapuptime is per-queue and therefore broken for any agent who is assigned to more than one queue), and neither form of Local (with or without /n) perform properly under this configuration. This is something new for me. Are you sure about this? Isn't wrapuptime taking in account agent state change? Because then, it's really bad for direct calls (for me it's rare that agent have several queues). Wrapuptime isn't a state in app_queue. You'd think it was (agent goes from in-use to in-wrapup, then back to available after the configured number of seconds). But what really happens is that the last time a member took a call is part of the member struct, which is subordinate to the queue struct (i.e. the last time a member took a call is stored per-member-per-queue, not per-member). You can test this easily by adding a member to two queues, then sending a call to each queue spaced a minute apart. Then run 'queue show' on the CLI. You'll see that the last was xxx seconds ago differs for the same member in the two queues. When app_queue is attempting to dequeue a call to an available agent, it checks if the current time minus the time of the last call for the member in this queue is less than the wrapuptime, and if it is, it skips that member and goes onto the next. Because the last call time is per queue, you can have an agent take a call on queue foo, hang up that call, and receive a call for queue bar one second later, even if both queue foo and queue bar have a wrapuptime of 60 seconds configured. So, the built in wrapup has three major problems compared to the ACD system we moved to Asterisk from: - wrapuptime is per-queue-per-member, not per-member - wrapuptime is invisible (the internal state of the member during the wrapup is not in use, and only by looking at the last was xxx seconds ago and knowing what the configured wrapuptime for the queue is can you tell that a member is not actually eligible to take a call) - wrapuptime cannot be shortened or extended by the agent (or anyone else for that matter - I once mused with writing a dialplan function to set the last call time to some arbitrary epoch value to make this viable) These problems led to me developing this external wrapuptime system. My original implementation for doing external auto-pause was to have an AMI client watch for Hangup events. When I did that, Local channels caused problems. If I used one form of Local channel (with or without the /n, I forget now) then the hangup event was fired when the call was bridged to the agent. The other way, the hangup didn't fire until the agent was done with the caller, but some equally unpleasant side effect manifested. This was all months ago, so I don't recall the specific problems I faced. The AMI solution was too CPU intensive, so I switched to a system that essentially tails queue_log. When it seems COMPLETEAGENT or COMPLETECALLER or TRANSFER, it uses AMI to pause the member, and schedules an unpause for x seconds in the future.By using PauseQueueMember without a queue name, it pauses an agent in all their queues, and if the agent needs to, he can unpause themselves early or cancel the pending unpause to give themselves more time to write notes. This way of doing things might not be incompatible with Local channels, but I'd have to do some tests. -- j. ___ 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 Died message
On 9/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: Or you might have two safe_asterisk processes trying to restart asterisk. A symptom of this (when Asterisk is not actively crashing) is constant remote UNIX connection messages on the console every few seconds (assuming you have nothing that legitimately polls Asterisk using 'asterisk -rx' running). The solution is to use ps to find out which of the safe_asterisk processes owns the actual running copy of Asterisk (using pid and ppid) and then kill the other one. -- j. ___ --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] Dialplan regexp
On 9/5/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Many thanks for that!! I didn't know that the order worked quite like
that but I see it now... Better go check the other contexts...
(the [56][0-9] worked fine).
You can also impose a finer level of control over the order extensions are
searched in by putting them in different contexts and using include to
pull them in in a specific order:
[foo]
exten = _017935201[56][0-9],1,Goto(local,${EXTEN:-3},1)
include = bar
[bar]
exten = _0.,1,Set(CALLERID(num)=${PSTN_GLOBAL}${CALLERID(num):-3})
exten = _0.,2,Dial(${TRUNK}/${EXTEN},,W)
Dialing 01793520158 would match the longer pattern in this case. The search
is done in the initial context, then in each included context in the order
they were included.
There's more info here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
--
j.
___
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] E1 to Ethernet Bridge
On 9/3/07, Arinze Izukanne [EMAIL PROTECTED] wrote: Can you show me a sample fo config? The link schematic should look like this: E1 == TDMoE==E1. Refer to the section Sample configs for setting up TDMoE between 2 servers without TDM hardware, using ztdummy on this page: http://www.voip-info.org/wiki/index.php?page=Asterisk+TDMoE The examples are for T1, so you'll have to change the number of channels on each span and change the channel numbers used for bchan= and dchan=, but if you're familiar with E1 deployment already this should be simple. -- j. ___ --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] Members in 'Unknown' status in output of 'queue show'
On 8/29/07, BJ Weschke [EMAIL PROTECTED] wrote: I think we will want to see what state chan_sip is sending into app_queue for it to be called Uknown. What is the last state these channels are in before they go to Unknown in app_queue? Unfortunately, I don't know. This is in an active call center (~20 calls going on at all times) and the first I hear about it is when agents start complaining of getting calls while already on a call. I'd have to add some debug code to both chan_sip and app_queue every time the state changes I expect, but the last time this happened was over 5 weeks ago and there's no obvious similarities between the two occurences. I'll add the debug code anyway just in case, but I don't think it will give me anything useful prior to opening the bug report. -- j. ___ --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 handle + prefix
On 8/30/07, Adrian Marsh [EMAIL PROTECTED] wrote:
[outgoing-pstn-international]
exten = _+.,1,Set(EXTEN=00${EXTEN:+1})
exten = _+.,2,NoOp(test line: ${EXTEN})
Setting ${EXTEN} won't work, but Goto(context,00${EXTEN:1},priority) will:
[foo]
exten = 7997,1,Answer
exten = 7997,n,Set(FOO=+1441793xx)
exten = 7997,n,Goto(foo,00${FOO:1},1)
exten = 7997,n,Hangup
exten = _0.,1,NoOp(${EXTEN})
exten = _0.,n,Hangup
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/427-9dd49740, )
in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/427-9dd49740,
FOO=+1441793xx) in new stack
-- Executing [EMAIL PROTECTED]:3] Goto(SIP/427-9dd49740,
foo|001441793xx|1) in new stack
-- Goto (foo,001441793xx,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/427-9dd49740,
001441793xx) in new stack
-- Executing [EMAIL PROTECTED]:2] Hangup(SIP/427-9dd49740, ) in
new stack
--
j.
___
--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] Monitor System using AGI Scripts
On 8/29/07, Nitesh Divecha [EMAIL PROTECTED] wrote: Basically, it would be a totally different system running Asterisk with AGI scripts and monitoring other systems (Web Servers, FTP, SMTP). Not specifically monitoring ports (80, 21, 25) but whole system. If system timeouts then AGI scripts are triggered and notify system admin. You'd be better to monitor using something like Nagios or one of the other open-source monitoring systems, then have the notification script (which should be customizable in your monitoring system) write a .call file to make Asterisk dial out and tell the sysadmin. To use the Asterisk dialplan to schedule and cycle checks of services . erm. no. -- j. ___ --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 Agents on Remote Asterisk server?
On 8/29/07, Aubrey Wells [EMAIL PROTECTED] wrote: I have a main Asterisk server, and a server at a branch location connected via a IAX2 trunk. I want to have a queue at the main location that has people from both locations as members. I got this working, but the trouble comes when the round-robin logic selects a member at the branch office to call. If that user is unavailable, their voicemail answers the call, and the main server detects this as an answered call and assumes the agent answered. This is obviously not what I want, as I would like for the call to roll to one of the other agents. Has anyone come across this before? Solutions? Don't contact the remote agents using a context that includes a call to VoiceMail(). Contact a remote context that dials the agent using Dial() with the appropriate timeout and hangs up if the agent is unavailable. Then app_queue () will do the right thing. -- j. ___ --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] Members in 'Unknown' status in output of 'queue show'
Does anyone know what can cause queue members to go into a status of Unknown? pbxtel-01*CLI queue show cshas 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442 (dynamic) (paused) (Unknown) has taken 2 calls (last was 101 secs ago) SIP/1440 (dynamic) (In use) has taken 2 calls (last was 3071 secs ago) SIP/1428 (dynamic) (paused) (Not in use) has taken 2 calls (last was 10818 secs ago) SIP/1404 (dynamic) (paused) (Not in use) has taken 2 calls (last was 2228 secs ago) SIP/1429 (dynamic) (paused) (Unknown) has taken 2 calls (last was 953 secs ago) SIP/1432 (dynamic) (Unavailable) has taken 5 calls (last was 1229 secs ago) SIP/1430 (dynamic) (In use) has taken 2 calls (last was 22744 secs ago) SIP/1435 (dynamic) (In use) has taken 3 calls (last was 13511 secs ago) SIP/1434 (dynamic) (Unknown) has taken 6 calls (last was 9504 secs ago) SIP/1424 (dynamic) (In use) has taken 4 calls (last was 16373 secs ago) SIP/1408 (dynamic) (paused) (Not in use) has taken 2 calls (last was 8685 secs ago) SIP/1203 (dynamic) (In use) has taken 3 calls (last was 16425 secs ago) SIP/1410 (dynamic) (Unknown) has taken 2 calls (last was 8629 secs ago) Callers: 1. Zap/50-1 (wait: 11:15, prio: 0) 2. Zap/36-1 (wait: 0:41, prio: 0) That's just one queue, but I had nearly all my agents just go into Unknown status. This is on * 1.4.10.1. I had this happen once in the past, but couldn't reproduce it in the lab. When this happens, 'ringinuse=no' stops working, because app_queue considers Unknown to be a valid state to dispatch a caller to. So my agents start getting flooded with calls while already on the phone, then the call-limit I've configured in sip.conf kicks in and my console fills up with this: pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22621]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22762]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to peer '1405' rejected due to usage limit of 2 pbxtel-01*CLI [Aug 29 16:44:04] ERROR[22686]: chan_sip.c:3169 update_call_counter: Call to peer '1410' rejected due to usage limit of 2 I had to restart Asterisk to clear the states - sip reloads, app_queue reloads didn't do anything. Any thoughts as to where to start debugging this? I killed * instead of stopping it so that I got a core file. There is nothing in the log to indicate what went wrong prior to the first instance of ...rejected due to usage limit. Anything else I should gather before submitting a bug? Thanks -- j. ___ --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] Distributed System
On 8/28/07, Philipp Kempgen [EMAIL PROTECTED] wrote: Realtime + MySQL does it. That needs some extra work but it's possible. Or DUNDi. JR just posted a quick tutorial on getting that up and running: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf -- j. ___ --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] Keeping queue counters after restarting
On 8/24/07, Marlon Dutra [EMAIL PROTECTED] wrote: Every queue has some status counters (completed, abandoned, hold time...) that are very useful for statistics. The problem is that those counters are reset every time Asterisk restarts. Is there a way to keep those counters, maybe in astdb? Also, is there a way to reset the counters through a cli command? Not sure about restarts, but trunk keeps them through reloads. How often are you restarting? From http://svn.digium.com/view/asterisk/trunk/CHANGES?revision=79638view=markup : Queue changes - * Added keepstats option to queues.conf which will keep queue statistics during a reload. I don't think there's a command to reset the counters - would be a good (and relatively simple I think) patch to offer up before 1.6 gets closer to a release. -- j. ___ --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] Stable-Stable Asterisk
On 8/24/07, Joshua Colp [EMAIL PROTECTED] wrote: I'm going to end this email with a question myself... how many people have Asterisk on a development/staging server before deployment, test, and isolate the issues they may have in their specific scenario? I do, but many of the problems I have experienced (see #10199 for an example) don't manifest under anything but production loads. In that particular case, I couldn't find a way to replicate the levels of traffic and the nuances of agent pickup / ignore / hangup / etc. in my lab. My current load test consists of a lab box generating about 50-75 concurrent calls to an ITSP that terminate on another * conencted to PRI. But what you do with a call when it hits your box can make a difference. I had a load test that just walked through my IVRs pressing random keys for about 5 minutes. I could load 4 PRI full of calls to that context and the box would be fine. The second I added queueing (so that there was SIP signalling out to agent softphones), I'd get a kernel panic. The agent didn't even have to pick up the phone - just making it ring was enough. Let me ask a question myself: what kind of regression test does * undergo before release, and what level of traffic gets put through stuff like app_queue? I assume it's not real-world scale, else these hard to pin down concurrency issues we're seeing would have been caught in test. -- j. ___ --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] 99 bottles of beer
On 8/21/07, Steve Edwards [EMAIL PROTECTED] wrote: To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... To control the power bar the Asterisk server is plugged into, press 5 click DAAD! The stupid phone isn't working! -- j. ___ --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] Rewriting the From and Subject from voicemail for a MMS Message to a Cell Phone - like visual voicemail
On 8/19/07, Steve Turner [EMAIL PROTECTED] wrote:
I would like to send Multimedia Messaging (MMS) email (gateway) to my
cell
phone and have the from and subject be the callerid/calleridnam
information
from the voice mail message.
voicemail.conf lets you change the from and subject line, and has
replacement tokens for ${VM_CALLERID}, ${VM_CIDNUM} and ${VM_CIDNAME}.
What are you trying to achieve that use of emailfrom, emailsubject and
attach=yes in voicemail conf won't do?
--
j.
___
--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] Suggestions on how to debug strange DTMF problems
I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems never occurred when we were using 1.2. They started immediately after upgrading to 1.4 (1.4.4 and 1.4.2.1 at the time IIRC) The original problem people would report is that they have problems using IVRs that they call over the PSTN. Duplicated digits, missed digits, etc. I've tried to replicate the problem using various user agents (Polycom 430, Grandstream GXP2000, Linksys SPA942, eyeBeam, etc.) and have observed the problem. It's not consistent, but that's probably due to the inconsistency of user input. If you concentrate very carefully and hit the keys for a consistent period of time with consistent spacing, the problem doesn't seem to happen. I also found that when the user agent was directed into DISA() and then dialed the IVR from within that application, they didn't have problems with their DTMF being recognized. In an attempt to quantify the problem, I set up the following test harness, sending calls out of Asterisk into an ITSP system. The final termination of the call is a different Asterisk box from the one that generated the call, though running the same version. DTMF from the ITSP to the second Asterisk system is SIP INFO. Asterisk - PRI (TE412P) - PSTN - ITSP - Asterisk I generated calls that executed SendDTMF(). On the other Asterisk system, I used Read() to capture the DTMF and stick it in a database. I did 1250 calls (250 to each of the 4 PRIs connected to the TE412P and 250 without regard to the PRI used). Across all of those calls, I didn't see one missed or doubled-up digit. I then did some manual tests where I placed calls from the Polycom 430 to the same application (basically putting myself in place of SendDTMF). Immediately I saw doubled up an missed digits. I turned on DTMF debugging on the second Asterisk box, and doubled-up digits always seem to take this form: [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 90 queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 70 ms [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 80 queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 50 ms [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 80 queued on SIP/5060-08da0d70 [Aug 17 10:50:05] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 That's a single keypress. As I understand it, synthesized DTMF tones should consistently have a duration of 100 ms, and indeed most of what shows up in the DTMF log does have that duration (or something close to it): [Aug 17 10:49:55] DTMF[25160] channel.c: DTMF end '1' received on SIP/5060-08da0d70, duration 100 ms [Aug 17 10:49:55] DTMF[25160] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/5060-08da0d70 [Aug 17 10:49:55] DTMF[25160] channel.c: DTMF end emulation of '1' queued on SIP/5060-08da0d70 [Aug 17 10:49:57] DTMF[25160] channel.c: DTMF end '2' received on SIP/5060-08da0d70, duration 100 ms [Aug 17 10:49:57] DTMF[25160] channel.c: DTMF begin emulation of '2' with duration 100 queued on SIP/5060-08da0d70 [Aug 17 10:49:57] DTMF[25160] channel.c: DTMF end emulation of '2' queued on SIP/5060-08da0d70 [Aug 17 10:49:58] DTMF[25160] channel.c: DTMF end '3' received on SIP/5060-08da0d70, duration 110 ms [Aug 17 10:49:58] DTMF[25160] channel.c: DTMF begin emulation of '3' with duration 110 queued on SIP/5060-08da0d70 [Aug 17 10:49:58] DTMF[25160] channel.c: DTMF end emulation of '3' queued on SIP/5060-08da0d70 [Aug 17 10:50:00] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 100 ms [Aug 17 10:50:00] DTMF[25160] channel.c: DTMF begin emulation of '4' with duration 100 queued on SIP/5060-08da0d70 [Aug 17 10:50:01] DTMF[25160] channel.c: DTMF end emulation of '4' queued on SIP/5060-08da0d70 [Aug 17 10:50:02] DTMF[25160] channel.c: DTMF end '4' received on SIP/5060-08da0d70, duration 100 ms Yet from time to time, the ITSP hears that supposed 100ms tone as two tones of 70 and 50ms, or 90 and 40, or some combination that makes it appear to be a doubled-up digit. I'd be tempted to say that the problem is whatever part of the ITSP that is interpreting the DTMF coming in from the PSTN, but the IVRs that my users complain about are operated by tons of different
Re: [asterisk-users] Experimenting- Sip dialing with Zap
On 8/16/07, John Meksavan [EMAIL PROTECTED] wrote:
CLI. What am I doing wrong? Thanks in advance.
The channel spec you need to use is:
Dial(Zap/g0/${EXTEN:1})
not
Dial({Zap/g0/{EXTEN:1})
Though bear in mind that the :1 is removing the first char of your
extension, so if you dial '123' on your Linksys, you'll dial '23' out your
analog line, which is unlikely to be what you want to do if said line is
connected to the PSTN. It's more typical to see something like
exten = _9NXXNXX,1,Dial(Zap/g0/${EXTEN:1})
which matches a 10 digit local number prefixed by nine, but removes the
leading 9 (using :1) because it's not needed (or wanted) by the telco.
--
j.
___
--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] IVR and MySQL
On 8/14/07, Atis [EMAIL PROTECTED] wrote: That's possible, but i wouldn't recommend on large production system. Using MySQL you would need to connect and disconnect all the time, and it takes resources.. I would suggest to append that info to CDR userfield (if you are storing your CDR in MySQL), and run periodically some script that extracts them. Of course it's more complex, but that would be my way. If the data you wish to store is more complex than stuffing in the CDR userfield would allow, you can always call out to an AGI which can write the data to whatever file format you want for later loading into a database. If you used FastAGI and a pre-forking AGI server model, you could even take the database connection hit when the AGI server starts. The per-call cost would then be the cost to establish the socket connection to the AGI server from Asterisk, the cost to perform the SQL inserts over an established database connection, plus whatever other calculation or transformation you needed to do before doing the insert. That architecture would hold up under a fairly large load. Perl's Asterisk::FastAGI framework lets you specify the number of pre-forked children to launch, plus you can tell each child to exit (spawning a replacement for the pool) after processing a certain number of transactions. It's very similar to the Apache prefork model. -- j. ___ --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] BLF with Aastra
On 8/14/07, Matt [EMAIL PROTECTED] wrote: I have a 536i expansion module attached to a 57i-CT. The BLF lights on the 536i will light up and work fine for a while... however after a bit they seem to loose their ability to see if someone is on a phone. They still work to dial, if I try to dial, however, they don't light up when someone makes a call, or if their phone rings. If I reboot the phone, the lights start working again (for a while). Does 'sip show subscriptions' indicate that the 57i is still subscribed to the extension for updates? If not, you might have to do a test with 'sip debug peer aastraname' to confirm that the subscription is being made properly on phone startup and not being removed by the phone in response to some state change. A quick glance at chan_sip.c indicates that if a user agent tries to subscribe with an expiry time greater than 'maxexpiry' from sip.conf(default 3600 seconds), the subscription expiry in Asterisk will be silently changed to whatever the allowed maximum is. So if the Aastra is trying to subscribe for say 3 hours and Asterisk doesn't allow subscriptions greater than one hour, then notify messages will stop being sent after one hour until the Aatra re-subscribes. I haven't delved in very deep, so I can't tell if the response to the UA indicates the actual expiry Asterisk used, but even so you'd have to be certain that the Aastra respects an expiry in the response that differs from what it asked for. When you're doing the debug (hopefully on a quiet system), watch the phone boot, then use 'sip show subscriptions' to get the call-id of the subscription. Then watch for console messages indicating that the call has been destroyed (which should come at the 1 hour mark or whatever time the Aastra used for it's subscription length. -- j. ___ --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] IVR and MySQL
On 8/14/07, Fabio Ardeola [EMAIL PROTECTED] wrote: Let say that the user entry during the call is a reference number of a house to rent. Would be possible to check if the reference number is a valid entry on the MySQL database and then base on its answer define the next menu item on the IVR menu. If you want to do something like that, you can either use the MYSQL function (with the attendant issues of connecting/reconnecting/etc.) or put all of the functionality in an AGI script. Since AGI can both receive information from and send commands to Asterisk, you can do pretty much anything you can code. There are programming frameworks for AGI for Perl, PHP, Java, and you could even do it in shellscript if you want. The communication channel between Asterisk and the script is stdin/stdout, so you're not restricted at all. Using AGI does make the the integrity of your system depend on an external component (i.e. if you're using FastAGI and the agi server goes down, your calls will just return immediately to the dialplan), but when you need to do something that doesn't fit intuitively into the Asterisk dialplan, I find it's the way to go. -- j. ___ --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] Hardware Platform Recommendations for Digium Card Compatability
On 8/10/07, Jason K. Carter [EMAIL PROTECTED] wrote: Could everyone that has a working production Asterisk server that uses a Digium telephony card as a BRI/PRI gateway let me know what motherboard/processor your server uses? Currently running a TE412P in a IBM x3650 Model 7979. I had some problems when I also had a TDM400B in the same system. I have also run this card successfully on a Intel SE7230NH-1 board (having the TDM400B installed as well was not a problem on this board) I had a reproduceable kernel panic under moderate load running this board on a HP DL380G5 with Zaptel 1.4. Zaptel 1.2 was just fine. All of my testing was done on CentOS 4.4 and 4.5. My zttest scores (on the IBM) are generally above 98%, but sometimes I see the tests start at 97.73% for about 20 seconds before it climbs. I often see spikes up to 100% rapidly followed by a drop back to 98.x%. -- j. ___ --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 Manager to Record Greetings
On 8/10/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: That's great, now say you have 5 or 6 AA's and each one has 10 different parts that you want to record (thank you for calling... for Steve press 1 for dave press 2). Rather than having to record a long message, I want to break it into pieces so that if dave leaves, we can just record that one chunk rather than the whole thing. I would need lots of extensions pre-setup for each chunk. Not very efficient. You could front it with something other than extensions. I tag all my recordings with a 4 digit number, but I use an AGI script to manage them. The AGI (written in Perl) authenticates the user, then lets them punch in the announcement number. A database lookup translates the announcement number to the pathname of the the file, then the user gets the choice to listen to the existing recording or re-record it. Adding new announcements is a simple SQL insert. TMTOWTDI -- j. ___ --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] generating a GUID
On 8/9/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
I have a need to have a GUID (for example,
bcd47ccc-d7c9-ddb6-dc11-6746a770d77d [36 characters long including the
-]) generated in the dialplan. Is there any asterisk function that
would do this ? I would prefer not to have to shell out every time a
call comes in.
There's nothing built in that I know of. I had mused with the idea of
wrapping the available UUID generator code out there into a function and
offering it as a patch, but it's a low priority thing for me.
In the meantime, you could achieve what you want without the cost of
spinning up a shell process by writing a FastAGI app in Perl. Using the
modules Asterisk::FastAGI and Data::UUID, you could get a UUID back for the
cost of the socket connection.
This is a quick example that I coded up to do that - it was actually more
painful to install the modules from CPAN than code up the server itself:
--START--
#!/usr/bin/perl
#
use strict;
use warnings;
MyAGI-run( port = 4574 );
package MyAGI;
use base 'Asterisk::FastAGI';
use strict;
use Data::UUID;
my $uuid;
sub child_init_hook
{
$uuid = Data::UUID-new;
}
sub fastagi_handler
{
my $self = shift;
$self-agi-set_variable( UUID = $uuid-create_str() );
}
---END---
When run, this creates a pre-forking server with 5 children, which makes the
individual UUID generation about as cheap as you're going to get going
outside of the Asterisk process. When I execute that with agi debugging
turned on from this diaplan snippet:
exten = 7993,1,Answer
exten = 7993,n,AGI(agi://127.0.0.1:4574/fastagi_handler)
exten = 7993,n,SayAlpha(${UUID})
exten = 7993,n,Hangup
I get this:
-- Executing [EMAIL PROTECTED]:1] Answer(SIP/427-9df490e0, )
in new stack
-- Executing [EMAIL PROTECTED]:2] AGI(SIP/427-9df490e0,
agi://127.0.0.1:4574/fastagi_handler) in new stack
AGI Tx agi_network: yes
AGI Tx agi_network_script: fastagi_handler
AGI Tx agi_request: agi://127.0.0.1:4574/fastagi_handler
AGI Tx agi_channel: SIP/427-9df490e0
AGI Tx agi_language: en
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1186667018.723
AGI Tx agi_callerid: 427
AGI Tx agi_calleridname: James FitzGibbon
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callington: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 7993
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: from-internal-admin
AGI Tx agi_extension: 7993
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx CLI
AGI Rx SET VARIABLE UUID 88AEDB9A-467E-11DC-9F13-8E31D47CEF85
AGI Tx 200 result=1
-- AGI Script agi://127.0.0.1:4574/fastagi_handler completed, returning
0
-- Executing [EMAIL PROTECTED]:3] SayAlpha(SIP/427-9df490e0,
88AEDB9A-467E-11DC-9F13-8E31D47CEF85) in new stack
And then Allison starts chattering out the digits of the UUID.
Hope that gives you something to work with.
--
j.
___
--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] Method for scripting options specified in make menuconfig
On 8/9/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: After you run make menuselect, you'll have a file 'menuselect.makeopts' in your asterisk source dir. Copy that to /etc/asterisk.makeopts (or ~/.asterisk.makeopts) and it will be used for future builds. Once you've copied the file over, do a 'make distclean ; ./configure ; make' to check that it worked. Hmmm why distclean ? 'clean' doesn't remove the generated menuselect.makeopts: clean: $(SUBDIRS_CLEAN) rm -f defaults.h rm -f include/asterisk/build.h rm -f include/asterisk/version.h @$(MAKE) -C menuselect clean cp -f .cleancount .lastclean distclean: clean @$(MAKE) -C menuselect dist-clean @$(MAKE) -C sounds dist-clean rm -f menuselect.makeopts makeopts menuselect-tree menuselect.makedeps rm -f makeopts.embed_rules rm -f config.log config.status rm -rf autom4te.cache rm -f include/asterisk/autoconfig.h rm -f include/asterisk/buildopts.h rm -rf doc/api rm -f build_tools/menuselect-deps So if you go through this cycle: untar ./configure make menuselect ...make module choices... cp menuselect.makeopts /etc/asterisk.makeopts make clean ./configure Then the automated run of menuselect is going to have two makeopts files that it might pull from: the generated one left over from the first run of configure, and the one in /etc. But since the files should be identical, you won't be absolutely sure that your file in /etc is the one driving the module choices. If you changed cp menuselect.makeopts... to mv menuselect.makeopts... in the above snippet, then I suppose 'make clean' would suffice. But 'make distclean' doesn't do any harm - it should return the directory to it's post-untar state, right? -- j. ___ --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] Need Help in changing Voice message
On 8/9/07, Farooq Ahmed [EMAIL PROTECTED] wrote: Asterisk has a lot of customizable voice prompt in /var/lib/asterisk/sound but i want to change a very well known voice message which occurs when we try to dail a number against dial plan beep beep beep The person you are calling is unavaiable, please try again. I thought it would be availabe in the sound directory of asterisk but it is not there. When i dial such wrong number no log appears in the asterisk cli command just get this message so i am not getting any idea which macro or application generating this message. Anybody have any idea about how to change this? This is probably not coming from Asterisk. It's probably generated by your phone when Asterisk responds with a 5xx or 4xx response code to your INVITE. Depending on your phone you may or may not be able to change it. -- j. ___ --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] Method for scripting options specified in make menuconfig
On 8/8/07, arkda [EMAIL PROTECTED] wrote: I've been digging around and I haven't found a way to do this, but I have a feeling I'll feel like an idiot because it's something I'm over looking. Normally if I need to specify an additional option (such as different language sound files) or I'm building an Asterisk server with a lean configuration and need to remove some modules I do so with 'make menuconfig'. I've ran into a need however to install Asterisk entirely from the command line, so I'm looking for the method of accomplishing what I've normally done through 'make menuconfig' solely from the command line. Anyone know how this is accomplished? After you run make menuselect, you'll have a file 'menuselect.makeopts' in your asterisk source dir. Copy that to /etc/asterisk.makeopts (or ~/.asterisk.makeopts) and it will be used for future builds. Once you've copied the file over, do a 'make distclean ; ./configure ; make' to check that it worked. It's the same idea for asterisk-addons, except you copy its menuselect.makeopts to /etc/asteriskaddons.makeopts. -- j. ___ --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 write a function with a return value in Asterisk
On 8/8/07, Mike [EMAIL PROTECTED] wrote: I'd be most thankful for some link to a page that shows how to write such a function in Asterisk. There is a test application in the source tree (not built by default I believe), but it doesn't look like anyone has made an equivalent sample function. However, many of the functions in 1.4 are pretty simple, and would be a good jumping off point. Take func_sha1.c for example: 83 lines in the file, 4 functions and one macro. You could copy that and do the proper renaming to make MY_FUNKY_NEW_FUNC that does exactly what func_sha1 does (or alternatively, nothing by getting rid of the bulk of the sha1() function therein. How big it gets as you add whatever magic that function should perform is up to you of course. -- j. ___ --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 write a function with a return value in Asterisk
On 8/8/07, Mike [EMAIL PROTECTED] wrote:
So, I wrote (well, plagarized directly from the Web) a simple Perl program
that prints Hello World. I call it using this:
exten = 12345,1,AGI(agi-helloworld.agi)
Seems to work (I'm not expecting anything, really, just no Asterisk
error).
When I try to use it as part Noop like this:
exten = 12345,1,Noop(${AGI(agi-helloworld.agi)})
In the hope of getting to see Noop(Hello World) in my CLI, I get the
following Asterisk error:
Aug 8 13:40:48 ERROR[5771]: pbx.c:1402 ast_func_read: Function AGI not
registered
AGI certainly seems registered as it worked in the first case. Again,
something obvious I missed?
This is calling the AGI application:
exten = something,priority,AGI(program|args)
This is an attempt to call a function called AGI (which doesn't exist) and
pass the results of said non-existent function to the NoOp application:
exten = something,priority,NoOp(${AGI(program|args)})
look at 'core show applications' and 'core show functions' to see what you
can call in each case. Applications and functions aren't interchangeable.
If you want to use an AGI script to set a variable you can later use as an
arg to Dial(), then you want to call the AGI application from your dialplan,
then from inside the AGI script do your calculations and issue the AGI
command SET VARIABLE name value.
So if you have a very basic AGI script that just does this:
echo SET VARIABLE foo bar
then your dialplan could look something like this:
exten = foo,1,AGI(foo.agi)
exten = foo,n,NoOp(${foo})
And you'd expect to see NoOp(bar) on your console when you called that
extension.
Of course, you'd want to use one of the available AGI frameworks to do the
heavy lifting of parsing the input that Asterisk gives an AGI script and
take care of the error handing when you issue a command back to Asterisk
from the AGI script.
--
j.
___
--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] Learn some terminalogy before mounting thistask.
On 8/5/07, James R. Stevens [EMAIL PROTECTED] wrote: In the design of an Asterisk system using Cisco 7900 series SIP phones we are struggling with giving the reception folks (3) hardware that can tell them the status of everyone in the office (10 or so) (On the phone, out of office etc) Something that would register each of the extensions we choose and give status of that ext. What hardware (Phone or other) could we give the receptionist to do this? You're probably looking for something like this: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008883d.html I have no experience integrating this specific piece of hardware with Asterisk, but I've done what you're trying to do with the Grandstream equivalent for our front reception: http://www.grandstream.com/gxp2000.html and http://www.grandstream.com/gxp2000ext.html As I understand it, so long as the device can do a SIP SUBSCRIBE for each extension you want to monitor and you configure hints in your Asterisk dialplan for those extensions, it should work. You may need to set 'subscribecontext' (in sip.conf) for the phone that will be watching the extensions unless your hints are in the same context as the phone uses for outbound dialing. Of course, what the device does with the various payloads contained in the SIP NOTIFY messages is going to be different for each phone. On the Grandstream I can see 'not in use' (solid green) and 'ringing' (flashing red) distinctly, but 'unavailable' and 'in use' are both mapped to a solid red, which makes it somewhat useless for transiently connected user agents like softphones. Hopefully someone with experience will speak up and confirm that the 7900 series does interop properly with Asterisk for SUBSCRIBE and NOTIFY. If that doesn't work, you could always go with a software solution, like the Flash Operator Panel. voip-info has a list (look at the Operator section on the page): http://www.voip-info.org/wiki/view/Asterisk+GUI -- j. ___ --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] IAX bat phone.
On 8/3/07, Michael Munger [EMAIL PROTECTED] wrote: Is there a way to setup an IAX bat phone (immediate=yes) or is this a privilege only reserved for ZAP channels? As I understand it, this would have to be supported by your specific hard/soft phone. It's the same with SIP - taking a handset off-hook doesn't cause any traffic to go to Asterisk. The first packet from the user agent is sent when the phone tries to dial something. Depending on the user agent, this could be as soon as someone presses a single key (so-called early dial with SIP 484 responses), or more typically when an entire number has been dialed and a timeout has occurred or send button has been pressed. Zap FXS ports can tell when a handset has gone off-hook and take some action based on that due to the change in electrical impedance. Some soft-phones support bat-phone operation, though you have to hunt through the docs to get it to work. My Linksys SPA942 desk phone has a dial plan syntax that allows this: (:S0) Which means prefix whatever I type with and match an empty string, dialing as soon as you have a match, which causes the phone to calll as soon as I take it off hook. But it's obviously device-specific, and has nothing to do with SIP or IAX or Asterisk for that matter. When the call arrives at my server, it doesn't look any different than a call to from a phone with a more traditional dialplan. -- j. ___ --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] Learn some terminalogy before mountingthistask.
On 8/6/07, James R. Stevens [EMAIL PROTECTED] wrote: I'm reading the PDF on the Cisco Expansion module and it says 'When used as a DN key buttons are illuminated …' Is that what we are doing within Asterisk or Trixbox when we configure an extension? (A Directory Number??) I suspect DN Key is just one way of describing a multi-function button that can both display extension status and serve as a speed dial / transfer destination. On the Grandstream I have to configure the expansion car buttons as Asterisk BLF buttons, even though BLF (busy lamp field) isn't an Asterisk setting that I turn on. To enable BLF functionality in Asterisk, I have to set up hints in the dialplan and configure the user agent to subscribe to status notitications for those extensions. I'd search for asterisk user testimonials to be safe (assuming nobody steps up and says I got that working). Often times you'll find someone's blog about how they got a feature working with a particular piece of hardware, along with configuration samples. -- j. ___ --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] Macro and Arguments
On 8/3/07, bilal ghayyad [EMAIL PROTECTED] wrote:
At the extensions.conf file, at [demo] context, there
is a line:
exten = 1234,n,Macro(stdexten, 1234,
${GLOBAL(CONSOLE)})
In this line, I understand that it calls the macro
name stdexten [macro-stdexten] but about the other
variables, do we consider 1234 is ARG1 and the
${GLOBAL(CONSOLE)} is the ARG2? This is important to
distinguish the arguments inside the macro.
Correct.
From the other side, why it used ${GLOBAL(CONSOLE)} to
retreive the variable and did not write it directly
${CONSOLE} as already CONSOLE is configured in the
[global] or what is the storey :) - ?
Using ${CONSOLE} relies on magic - it looks for a channel variable, and when
one is not found, it falls back to the global var. If CONSOLE happened to
be defined on the channel, it would be returned instead of the global var.
Using the GLOBAL dialplan function lets you get at global variables whether
or not direct access to them is occluded by a like-named local or channel
variable.
--
j.
___
--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] DIALSTATUS not set
On 8/3/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: why if I call the Busy or Congestion extensions, the DIALSTATUS and HANGUPCAUSE variables are not set ? If I call (say) extension 1234 all things are set ok. I think you've answered your own question there. The only asterisk application that sets DIALSTATUS is Dial(). If you grep the source, you'll see that the value is retrieved by some other modules (chan_sip, chan_iax, etc.), but only Dial() sets the value of the variable. I assume when you say when I call the Busy extension you mean something like a SIP user whose context is outgoing doing an INVITE to [EMAIL PROTECTED]. If so, you're bridging a SIP call leg to an asterisk application, so Dial() isn't invoked and DIALSTATUS isn't set. It might work if you did an invite to an extension that used Dial() to call a Local channel (e.g. Local/[EMAIL PROTECTED]), but I'm not sure how DIALSTATUS would interact with the /n option on the local channel. -- j. ___ --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] partial ChanSpy
On 8/3/07, nik600 [EMAIL PROTECTED] wrote: is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. trunk has added an 'o' option to ChanSpy: o - Only listen to audio coming from this channel.\n You might be able to achieve what you want by alternately spying on either side of the bridged call using 'o' both times. I'm not sure if this would be portable back into 1.4 though or if you'll have to wait for 1.6 to be released. -- j. ___ --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] Hardware that can ring my phone?
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box I tend agree with your evaluation. Still, I was thinking that since all these el-cheapo SOHO PBX boxes support manual attendant call transfer, what's to prevent Asterisk from mimicking an attendant by sending proper DTMF signals and make this box transfer the call to the single analog phone in the business? That is, Asterisk will connect (via RJ-11) to the unit as the attendant's phone, and my real phone (only one in the system) will connect via a second RJ-11 (there could be 4 of them). Or is Asterisk not capable of sending DTMF signals over an RJ-11 connection? You can send arbitrary DTMF over any of Asterisk's channels from the dialplan. I just figured that this level of integration was a bit deeper than you were looking for as a first project. It would be an interesting experiment, to be sure. The biggest issue I'd think would be feedback - you can send the DTMF along the wire, but how do you know that the SOHO box interpreted it correctly? If the only feedback is designed for a human (i.e. auditory), then interpreting those cues with Asterisk would be non-trivial. Do I undestand correctly that with this solution, I will still be able to connect to my analog Verizon phone line with the SIP phone? That is, the outside world will see my phone as an ordinary phone, when in fact I am using a SIP phone? If so, that means that Asterisk does all the magic behind the scene, right? Yes, your Verizon POTS line would go into a FXO port in your server (which in Asterisk would be referenced as the channel Zap/1 - zaptel being Asterisk's TDM driver) and your SIP phone would connect via your standard office network and be referenced as SIP/whateverusernameyouwant. A very simplistic example of bridging a call would be: [from-verizon] exten = s,1,Dial(SIP/whateverusername) Assuming that you'd configured zaptel to route calls that come in on the FXO port to the Asterisk context named from-verizon, then any such calls would immediately cause Asterisk to ring your SIP phone, and if answered to bridge the two calls together. A more complex example that makes them press one to call you and otherwise lets them leave a message: [from-verizon] exten = s,1,Background(Press1ToTalkOr2ToLeaveAMessage) exten = s,n,WaitExten(10) ; timeout exten = t,1,Goto(vm,1) ; invalid exten = i,1,Goto(vm,1) ; press 1 exten = 1,1,Dial(SIP/101,20) exten = 1,n,Goto(vm,1) ; press 2 exten = 2,1,Goto(vm,1) ; all voicemail activity ends up here exten = vm,1,VoiceMail(u101) exten = vm,n,Hangup [from-officephone] exten = *98,1,VoiceMailMain extne = *98,n,Hangup Assuming you've now set up your SIP phone as extension 101, this would play a sound file saying press 1 to talk to 2 to leave a message. If they press 1, your SIP phone rings. If they press 2, they go to voicemail. If they wait 10 seconds without pressing anything, or press something other than 1 or 2, they also go to voicemail. If they press 1 to dial your phone and you don't pick up after 20 seconds, they go to voicemail. On your deskphone (could just as easily be a SIP softphone if you prefer), you can dial *98 to log in and pick up your new voicemail messages. Hope that demystifies some of what you're trying to do. -- j. ___ --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] Recording calls after queues?
On 8/2/07, Jay Moore [EMAIL PROTECTED] wrote: With my current setup, I record all incoming calls to my queues. My problem is that once a call is transferred out of a queue, recording stops. How can I make it so recording continues even after a call is transferred? If you need me to post any dialplan or conf logic, please ask. Explicity invoke MixMontitor() in your dialplan before calling Queue() instead of using monitor-format=whatever in queues.conf. If you get to some point where you want to stop the recording, call StopMixMonitor(). -- j. ___ --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] TE120P in Canada
On 7/31/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: span=1,1,0,ccs,hdb3,crc4 I was under the impression that ccs/hdb3 was more typical of E1 service than T1. I ran across this when looking up something on span syntax yesterday (from http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax): Framing= how to communicate with the hardware at the other end of the line. For T1: Framing is one of *d4* or *esf*. For E1: Framing in one of *cas* or *ccs*. Coding= another parameter of the communication with the other end of line hardware. For T1: coding is one of *ami* or *b8zs* For E1: coding is one of *ami* or *hdb3* (E1 may also need crc4) I have NI-2 PRI service from Telus in Ontario, and my spans are set up as: span=1,1,0,esf,b8zs And my zapata.conf reads: switchtype=national signalling=pri_cpe I don't have rxwink explicitly set to anything. I'm not a digital trunk expert by any means, but I thought the wink/flash/start time settings were used on trunks that don't have a dedicated signaling channel the way PRI does. If you leave it as pri_net, you'll probably see messages to the effect that I think I'm PRI_NET but so does the other end. -- j. ___ --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] Hardware that can ring my phone?
On 8/1/07, Linux Lover [EMAIL PROTECTED] wrote: But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? This SOHO PBX box won't interop with Asterisk because it doesn't speak any of the protocols that Asterisk does. This box appears to be a solid-state (and I'd assume very feature restricted) alternative to Asterisk. That it happens to have both FXO (to the Telco) and FXS (to the analog phone) ports doesn't mean that it is usable as an analog interface for Asterisk. Your best bet is to find your closest Asterisk user's group and see when they're next doing a build seminar. Most user groups do these a few times a year and you might be able to find someone who will do one on demand. You bring some cheapo PC you might have lying around and buy a $20 FXO card and build a simple answering machine using Asterisk. From there, it's easy to extend so that when the user chooses a particular option in your IVR, the call is bridged to a phone in your office. The original single-FXO-port card from Digium was the X100P. These aren't sold anymore (the TDM400B modular card replaced it), but they can be found on eBay for $10-$30. If you can get your hands on one, you might consider going with a cheap SIP phone instead of a analog phone for your business. There isn't (as far as I know) a readily available cheap single-FXS-port card. If you go with an analog phone behind Asterisk, you'll need an FXS port. If you go with a SIP phone, you just need to have a network connection from the phone to the server, which might be cheaper. A quick search on eBay shows a few Grandstream Budgetone 101 phones (certainly not the best available, but they'll do the job) in the sub-$50 range. -- j. ___ --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 on 64-bit?
On 7/31/07, Benjamin Jacob [EMAIL PROTECTED] wrote: Searched all over, but couldn't find anything conclusive. Does an off-the-shelf version of Asterisk run without any issues on a 64-bit machine? Does anyone have any 'conclusive' figures? I've run 1.2.14 - 1.2.18 and 1.4.4 - 1.4.9 on CentOS 4.4 and 4.5 x86_64 with no problems. If your distro is one of those supported by http://www.atrpms.net/ then you should be fine - they package zaptel and asterisk for x86_64. If not, it works fine to build on your own, though some versions are a bit finicky (IIRC, earlier versions of 1.2.x wouldn't look for libpri in /usr/lib64, just in /usr/lib. That resulted in a chan_zap module without PRI support if you didn't modify the configure script. 1.4 doesn't seem to suffer from these problems, as they've revamped the whole build system since 1.2. -- j. ___ --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] Problems using TE412P and TDM400B in a IBM x3650
Another day, another apparant unexplained hardware incompatibility. I have a TE412P and a TDM400B living quite happily in a whitebox using an Intel motherboard: http://www.intel.com/design/servers/boards/se7230nh1-e/index.htm I tried to move to an IBM x3650 system. It uses a slightly newer chipset, but apparantly it's in the same family. The SE-7230 board has been EOL'd and the suggested replacement uses the same chipset as the x3650. I had to get a PCI-X riser cage to put the cards into, as the server only supports PCIe as shipped. http://www-03.ibm.com/systems/x/rack/x3650/specs.html When I just have the TE412P in the server, no problems. If I put both the TE412P and the TDM400B in, I get no end of errors. When I put the TE412P in the first PCI-X slot and the TDM400B in the second, then none of my PRI channels will get out of red alarm - they go red as soon as I load zaptel, and stay there through ztcfg, starting asterisk, restarting zaptel via the Asterisk CLI, etc. If I swap the cards, then only one of the ports (#4) stays in red alarm, while the other 3 seem to be fine. I checked /proc/interrupts, and both cards were getting their own interrupt (forgot to save the output unfortunately, and I'm back on the original hardware right now). Has anyone run this type of hardware combo successfully, or had similar problems on other hardware that they got around? Thanks -- j. ___ --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] Queues with logged in agents that are not reachable
On 7/30/07, voiplist [EMAIL PROTECTED] wrote:
I noticed that if I have an agent logged in using AgentCallBackLogin
and that agent is unreachable for some reason (SIP phone unplugged)
calls to him/her will completely yack.
For example:
1-Agent 500 is the only one logged into queue number 1.
2-A call comes into queue number 1
3-The call is pushed to agent 500 at extension 21 which is unreachable
because the ethernet cable is unplugged to extension 21's handset.
4-The caller gets hungup on entirely instead of the call going to
another agent or leaving the caller in the queue
I don't expect this to happen but I want to be sure all bases are
covered on light days during shift changes etc.
This is either a problem with your dialplan or your queue configuration.
If you always want your callers to enqueue regardless of agent status, make
sure that joinempty=yes and leavewhenempty=no in queues.conf for that
queue. You may also want to add a
exten = whatever,n,NoOp(${QUEUESTATUS})
right after your call to Queue() to see why the calls are leaving the
dialplan. I suspect that you've got one or the other of those settings not
set properly, so when there are no available agents, your calls exit the
Queue() application with $QUEUESTATUS set to JOINEMPTY or LEAVEEMPTY,
but you don't have anything in your dialplan following Queue(), so they run
off the end of the extension and * hangs up on them.
Note that there is a problem with 1.4.9 that breaks
joinempty=yes/leavewhenempty=no - there's a patch offered to my bug report (
http://bugs.digium.com/view.php?id=10320), but due to other strange
instability observed in 1.4.8 and 1.4.9, I'm back on 1.47.1, so I haven't
tested it yet.
--
j.
___
--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] Problems with new logic being 'n' option to Queue in 1.4.9
On 7/27/07, James FitzGibbon [EMAIL PROTECTED] wrote: I'll go open a bug report. http://bugs.digium.com/view.php?id=10320 For anyone who wants to track it. -- j. ___ --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] Problems with new logic being 'n' option to Queue in 1.4.9
On 7/26/07, James FitzGibbon [EMAIL PROTECTED] wrote: Is it possible for qe.parent-membercount to be set to zero in a queue where all agents but one are on the phone and that one remaining agent lets their phone ring without answering it? I added some debug code to app_queue and ran a few tests. The change in app_queue in 1.4.9 breaks queues configured as joinempty=yes. If there are no members in the queue, the membercount will be 0. A queue configured as joinempty=yes should still allow calls to be enqueued in this case. Because go_on is zero and membercount is zero and the comparison is go_on = qe.parent-membercount, any calls that attempt to join a queue without any members will immediately get kicked back to the dialplan. -- Executing [EMAIL PROTECTED]:16] Queue(Zap/23-1, X|tW) in new stack -- Started music on hold, class 'default', on Zap/23-1 [Jul 27 07:49:27] WARNING[31209]: app_queue.c:3458 queue_exec: about to compare X's go_on (0) = qe.parent-membercount (0) -- Exiting on time-out cycle Though I can't seem to lock onto the circumstances, I also oberved this breaking queues configured as leavewhenempty=no. I can't seem to replicate it using a single queue and a single member (on pause or on a call or letting their phone ring through), but as I described in my first message, I did see the Exiting on time-out cycle message last night when I had 4 people in the queue, all on a call, and one person didn't pick up their phone. Still, the joinwhenempty breakage should be enough to prompt a backout or rapid fix for this. Until recently, my dialplan didn't gracefully handle a return from Queue() because all my queues were configured as joinempty=yes and leavewhenempty=no. Recently I added logic to kick users back to the IVR if Queue returns for any reason, but had I not made that change, 1.4.9would have resulted in my callers getting unceremoniously hung up on if they tried to join a queue without agents. I'll go open a bug report. -- j. ___ --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] Problems with new logic being 'n' option to Queue in 1.4.9
On 7/27/07, Mark Michelson [EMAIL PROTECTED] wrote: Could you submit this as an issue on the bugtracker? The 'n' option was mucked with just prior to the 1.4.9 release and so any problems experienced with it should be reported there so they can be fixed as quickly as possible. It's been submitted; to clarify though the change made in 1.4.8 related to the 'n' option doesn't seem to cause this. 1.4.8 (with or without the 'n' option) does what I'd expect it to. It's the logical overloading of go_on in 1.4.9 that broke things. go_on used to be a boolean - now it's a dual-valued variable. Sometimes it's compared just for it's truthfulness, and sometimes it's used in a numeric comparison against the queues's membercount. If go_on had been left alone and a new variable used to store the threshold to be compared against membercount, I don't think 1.4.9 would have broken things, because go_on would (in my configuration) never have been set to anything but zero, preventing the numeric comparison against membercount from ever being performed. -- j. ___ --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] Problems with new logic being 'n' option to Queue in 1.4.9
I am experiencing a change in behaviour of my Queues in 1.4.9 vs 1.4.8.
I do not pass the 'n' option to any call to Queue() in my dialplan. Yet
since I upgraded to 1.4.9, I have occasionally seen this on my console:
-- Nobody picked up in 2 ms
-- Exiting on time-out cycle
That log message Exiting on time-out cycle is exclusive to the logic in
app_queue meant to deal with the 'n' option. If you don't pass 'n', you
should never see it.
1.4.8 code:
/* exit after 'timeout' cycle if 'n' option
enabled */
if (go_on) {
if (option_verbose 2)
ast_verbose(VERBOSE_PREFIX_3
Exiting on time-out cycle\n);
ast_queue_log(args.queuename,
chan-uniqueid, NONE, EXITWITHTIMEOUT, %d, qe.pos);
record_abandoned(qe);
reason = QUEUE_TIMEOUT;
res = 0;
break;
}
1.4.9 code:
/* exit after 'timeout' cycle if 'n' option enabled
*/
if (go_on = qe.parent-membercount) {
if (option_verbose 2)
ast_verbose(VERBOSE_PREFIX_3
Exiting on time-out cycle\n);
ast_queue_log(args.queuename,
chan-uniqueid, NONE, EXITWITHTIMEOUT, %d, qe.pos);
record_abandoned(qe);
reason = QUEUE_TIMEOUT;
res = 0;
break;
}
In both versions, the variable 'go_on' starts off set to 0, and only gets
set if you pass the 'n' option to Queue(). The manner in which it gets set
differs between 1.4.8 and 1.4.9, but it is only when you pass the 'n'
option, so it shouldn't matter. In my configuration, go_on should always be
zero.
The logic check around go_on is what's worrying me. In 1.4.8, go_on had one
of two values - 0 or 1. If you never passed 'n' to Queue(), it was always
0, so the block of code that takes you back to the dialplan on timeout can
never be executed.
In 1.4.9, if qe.parent-membercount is zero and you didn't pass the 'n'
switch, then you'll exit the queue as if you had timed out, even though you
never passed the 'n' option. I haven't gone through the entire code of
app_queue to see exactly how membercount gets manipulated, but it seems from
my log that these exitwithtimeouts events seem to occur right after an agent
has let their phone ring without picking it up (see the nobody picked up in
2ms message in my example above).
Is it possible for qe.parent-membercount to be set to zero in a queue where
all agents but one are on the phone and that one remaining agent lets their
phone ring without answering it?
--
j.
___
--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] vm-duration announcement missing?
I just saw this on my console:
[Jul 26 11:36:30] WARNING[8667] file.c: File vm-duration does not exist in
any format
[Jul 26 11:36:30] WARNING[8667] file.c: Unable to open vm-duration (format
0x4 (ulaw)): No such file or directory
Thinking I might have lost a file during a fsck or something, I checked -
sure enough, there's no file vm-duration in any format. I downloaded the
current (as of June 14th) core and extra sounds, but it's not in there
either.
1.2.x didn't use this file, but app_voicemail contains reference to it in
1.4.x - as far back as 1.4.0:
if ((!res) (durationm = minduration)) {
res = wait_file2(chan, vms, vm-duration);
[snip stuff about polish syntax]
res = ast_say_number(chan, durationm, AST_DIGIT_ANY,
chan-language, NULL);
res = wait_file2(chan, vms, vm-minutes);
}
Does anyone know where this file can be fetched from, or at least what it's
supposed to say?
Looking back at my logs, there are semi-regular instances of this error
message. In a default setup, it's only used if the message is more than 2
minutes long, which I guess most of my user's VMs aren't.
Thanks
--
j.
___
--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] Wake-Up Call didn't work
On 7/24/07, Asterisk guy [EMAIL PROTECTED] wrote: -- Attempting call on Local/[EMAIL PROTECTED] for application MusicOnHold() (Retry 1) Jul 24 08:23:17 NOTICE[21177]: chan_local.c:479 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Jul 24 08:23:17 NOTICE[21177]: channel.c:2409 __ast_request_and_dial: Unable to request channel Local/[EMAIL PROTECTED] ( but i have a extension 6009 login to * ) , what is the problem? Regardless of what endpoints you may have registering to your *, your dialplan does not allow that endpoint to be reached via extension 6009 in the 'default' context. Look at the file that gets put in outgoing (comment out the rename in the AGI script so it stays in /tmp. Then go read up on call files on the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Calls generated by call files need to have a starting point and a destination. The starting point for stuff like this is typically a Local channel, and the destination is either a context/extension/priority or an application with arguments. Either your starting point or your destination is invalid. IMO, skip the AGI for now. Get the file that your AGI is writting to /tmp and make a copy of it. Modify the copy, then move it to the outgoing dir and see what happens. If it doesn't work, make more changes. Without seeing your dialplan or the callfile, we can't diagnose your problem, but the error messages are pretty informative as to what asterisk was trying to do. Once you've successfully generated a call manually, then go back to having your AGI try to generate them automatically. -- j. ___ --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] MySQL components in asterisk-addons not being built
On 7/24/07, hugolivude [EMAIL PROTECTED] wrote: Thanks or all your help! I've posted the ./configure output below. I noticed that it says: checking for mysql_init in -lmysqlclient... no Presumably that's a problem, but I don't know how to fix it!! As I mentioned, I have MySQL installed and it works fine. rpm -qa indicates: MySQL-server-5.0.22-0 MySQL-devel-5.0.22-0 MySQL-client-5.0.22-0 How do I get mysql_init set up properly, if indeed that is the source of my problem? Post the snippet of config.log that deals with mysql, as that will give more detail as to why it's not finding mysql_init. For example, mine has this: configure:6161: checking for mysql_config configure:6179: found /usr/bin/mysql_config configure:6191: result: /usr/bin/mysql_config configure:6223: checking for mysql_init in -lmysqlclient configure:6258: gcc -o conftest -g -O2 conftest.c -lmysqlclient -L/usr/lib64/mysql -lmysqlclient -lz -lcrypt -lnsl -lm -L/usr/lib64 -lssl -lcrypto 5 configure:6264: $? = 0 configure:6282: result: yes Yours will likely have several iterations of trying to find it, indicating that the script is looking for the mysql libs in several directories and then when it fails to find it, gives up. Mine is only one iteration because it was found in the first place the configure script looked. Also, post the output of rpm -ql MySQL-devel | grep client and rpm -ql MySQL-client | grep client from the looks of the RPM package names, you aren't running the same distro as me (CentOS), but I suspect that the problem is that your RPMs have stuck the libraries in a non-standard place that the asterisk-addons configure script doesn't know to look in. Once you've figured out what that non-standard place is, it should be a simple matter of passing --with-mysqlclient=PATH to ./configure to make it look for your libs in their actual home. -- j. ___ --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] Wake-Up Call didn't work
On 7/23/07, Dovid B [EMAIL PROTECTED] wrote: Can it be that asterisk does not have permission to copy the file over ? Also check your date settings on the server. Yes, it's interesting that the page intro includes the sentence Lots of error checking to make sure its done correctly, but the final step that makes the process work (ensuring that the callfile ends up in the directory that pbx_spool is watching) doesn't have any error checking: touch( $wakefile, $time_wakeup, $time_wakeup ); rename( $wakefile, $callfile ); The fact that you see files in /tmp when all is said and done means that at least some of the script is working. A few things to check: Do the files in /tmp have the correct timestamp (file matches the requested wakeup time)? If so, then everything preceeding the rename seems to have worked, so check if the user running the AGI can move files from /tmp to /var/spool/asterisk/outgoing. Though given that it's an AGI being run by *, you'd have to have a pretty strange setup for that to fail. Perhaps the outgoing directory just doesn't exist (was never created for some reason?) If the files don't have the correct timestamp, start following the logic backwards. Do they look complete? Look through the AGI for places where the wakeup file is written to (i.e. fputs( $wuc, maxretries: $parm_maxretries\n); ) and check that everything that should be written is being written Working backwards you should be able to figure out where the script is failing, then you can check everything that comes afterwards as the user running the AGI to make sure that permissions and directories are set up properly. -- j. ___ --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] priorityjumping not working, Dial goes to n+1 not n+101
On 7/20/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Did you read UPGRADE.txt? Priority jumping was deprecated in 1.2. I assume it was removed from 1.4. According to UPGRADE.txt, the default in the absence of priorityjumping= changed from yes in 1.2 to no in 1.4: * In previous Asterisk releases, many applications would jump to priority n+101 to indicate some kind of status or error condition. This functionality was marked deprecated in Asterisk 1.2. An option to disable it was provided with the default value set to 'on'. The default value for the global priority jumping option is now 'off'. But there is no indication that they removed the option to have priorityjumping turned on globally. -- j. ___ --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] Zaptel 1.2.19 and 1.4.4 released
On 7/16/07, The Asterisk Development Team [EMAIL PROTECTED] wrote: fix various known issues. See the ChangeLog included in the releases for a full list of changes. The ChangeLogs are also available separately on the ftp site. Is there any more information available on this change? 2007-07-13 08:22 + [r2733-2736] Tzafrir Cohen [EMAIL PROTECTED] * Fix a digit mapping bug with hardware dtmf detection (r4357) I assume those revision numbers (4xxx) in the changelog are from another SVN repo - I was trying to figure out what exactly had changed, but the large number of files changed across 3 revs makes it a bit tough to isolate. I'm fighting with a DTMF problem right now where directly dialed SIP-Zap (PRI) calls produce doubled-up DTMF on the other side (not every digit, but just one causes the remote IVR to go into fits). The problem first appeared when we moved from 1.2.x to * 1.4.4/Zaptel 1.4.3. I haven't had much time to investigate - it happens whether the SIP agent connected to * is using inband, rfc2833 or info and across Grandstream, Aastra and Polycom units. Interestingly, if I send the SIP phone into DISA() pointing at it's own context, I get no DTMF problems on the remote side. Thanks -- j. ___ --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] Cisco 7940 log on/off
On 7/16/07, Adrian Marsh [EMAIL PROTECTED] wrote:
Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?
My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could have different users sitting there. Is
there any concept of logging on in these environments?
You can set up single-person queues that you can use like extensions. The
deskphone itself would still have to be addressable as SIP/something, and
the outbound context will be the same regardless of who is logged in (unless
you want to get fancy and store who is logged in in ASTdb then fork to the
proper outbound context using Goto in the shared context for the peer), but
it works.
Here's my queues.conf for a single-person queue:
[queuename]
strategy=rrmemory
servicelevel=60
timeout=20
retry=5
wrapuptime=0
maxlen=1
announce-frequency=0
announce-holdtime=no
joinempty=strict
leavewhenempty=strict
reportholdtime=no
monitor-format=wav
ringinuse=no
And the dialplan to enqueue:
--START--
[somecontext]
exten = ###,n,Macro(singlequeue,queuename,###)
[macro-singlequeue]
; No-op to give us a 1 priority
exten = s,1,NoOp
; if DIAL_ANNOUNCE is set, play the please hold on while I... msg
exten = s,n,GotoIf($[${DIAL_ANNOUNCE} != 1]?check_clid_set)
exten = s,n,Playback(transfer)
; Set CLID Name if specified
exten = s,n(check_clid_set),GotoIf(${ISNULL(${CLID_NAME})}?enqueue)
exten = s,n,Set(CALLERID(name)=${CLID_NAME})
; Enqueue the caller
exten = s,n(enqueue),Queue(${ARG1}|nrtW|||20)
; Jump based on status - TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY |
JOINUNAVAIL |
LEAVEUNAVAIL
exten = s,n(branch),Goto(s-${QUEUESTATUS},1)
; If unavailable, send to voicemail w/ unavail announce
exten = s-TIMEOUT,1(vmu),Voicemail(${ARG2}|uj)
exten = s-TIMEOUT,n,Hangup
exten = s-TIMEOUT,vmu+101,Playback(vm-theperson)
exten = s-TIMEOUT,n,SayDigits(${ARG2})
exten = s-TIMEOUT,n,Playback(vm-isunavail)
exten = s-TIMEOUT,n,Hangup
; If busy, send to voicemail w/ busy announce
exten = s-FULL,1(vmb),Voicemail(${ARG2}|bj)
exten = s-FULL,n,Hangup
exten = s-FULL,vmb+101,Playback(vm-theperson)
exten = s-FULL,n,SayDigits(${ARG2})
exten = s-FULL,n,Playback(vm-isunavail)
exten = s-FULL,n,Hangup
; Treat the empty, unavail as full
exten = s-JOINEMPTY,1,Goto(s-FULL,1)
exten = s-LEAVEEMPTY,1,Goto(s-FULL,1)
exten = s-JOINUNAVAIL,1,Goto(s-FULL,1)
exten = s-LEAVEUNAVAIL,1,Goto(s-FULL,1)
; Treat anything else as timeout
exten = _s-.,1,Goto(s-NOANSWER,1)
; if people star out of voicemail, send them to the top-level admin IVR
exten = a,1,GotoIf(${ISNULL(${IVR_CONTEXT})}?setivrcontext:gotoivr)
exten = a,n(setivrcontext),Set(IVR_CONTEXT=ivr-admin)
exten = a,n(gotoivr),Goto(${IVR_CONTEXT},s,1)
; If they hit 0 from VoiceMail to dial the operator,
; after 20 seconds they go back to the top of the named context
exten = o,1,Macro(operator)
exten = h,1,Macro(loghangupcause)
---END---
There's some extra cruft in there you could easily cut out to suit your
environment. Using that macro in the same way I would normally use Dial,
the experience for both the caller and agent is pretty much the same. You
need to have 1.4.7 so that if the agent hits ignore or reject the call
immediately exits Queue() - there was a bug surrounding the 'n' option in
1.4.6 and earlier. If you don't have that version, then hitting ignore or
reject will send the call back to the phone after 'retry' seconds until the
timeout runs out (in my case 20 seconds).
All you then need is some dialplan sugar to invoke AddQueueMember and
RemoveQueueMember in response to logon and logoff actions by the agent. If
you're not worried about authenticating the users, you can just do something
like:
exten =
_7000XXX,1,AddQueueMember(singlequeue_${EXTEN:4},${CUT(CHANNEL,,1)},,,${CUT(CHANNEL,,1)})
exten =
_7001XXX,1,RemoveQueueMember(singlequeue_${EXTEN:4},${CUT(CHANNEL,,1)},,,${CUT(CHANNEL,,1)})
Then agent 123 can log in by dialing extension 7000123 and log off by
dialing extension 7001123. If you attempt to enqueue the caller to the
queue when there's nobody logged in, then you'll immediately get kicked to
your dialplan with ${QUEUESTATUS} set appropriately. This may or may not be
what you want, because it means that callers will go directly to voicemail
if the person they're calling isn't logged in.
HTH
--
j.
___
--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] Channel name in queue log replaced by a manager event?
On 7/12/07, Jared Smith [EMAIL PROTECTED] wrote: It probably wouldn't hurt to open a bug for this... I've seen something like this before, only it was manager events ending up inside of SIP traffic. It definitely sounds like a pointer problem or maybe a locking problem to me... which means it's probably going to be difficult to track down. Filed as 10199, with a bit more info about the queue config and dialplan being used to enqueue callers. Thanks -- j. ___ --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] Queues monitoring software
On 7/12/07, Stefan Reuter [EMAIL PROTECTED] wrote: You might want to have a look at QueueMetrics: http://queuemetrics.loway.it/ I am not sure if it supports all features you are looking for but it should be a good start. QueueMetrics is working well for me in a 75 seat call center, but it won't do everything you ask for. - It doesn't have an interface to let you manage your Asterisk Configuration (though it can pull info from your Asterisk configuration apparantly - I opted to set things up manually) - it has a realtime view that uses the Manager API, but this is to show queue status, not other aspects of your * operation (like Zap status) Other than that, it's relatively easy to set up. A tad annoying if your * and QM live on different boxes, and I regularly lament the inability to modify the output much (all of the calculations and most of the output are tied up in compiled Java classes), but it took care of 80% or so of my reporting needs in a few days when the alternative was to roll my own reporting system. A combination of QM plus some homegrown stuff, or maybe QM plus one of the other Asterisk Management web portals might do the trick for you. -- j. ___ --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] Channel name in queue log replaced by a manager event?
On 7/6/07, James FitzGibbon [EMAIL PROTECTED] wrote: Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: Nobody else seeing this? I'm at a loss - it's only one queue now that I go and look at the history, but that queue is not configured differently from any other queue in the system. I've since moved from 1.4.6 to 1.4.7.1 and it's still happening. Should I just open a bugreport without any reliable way to reproduce the bug? -- j. ___ --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] Monitor events?
On 7/9/07, Daniel Gradecak [EMAIL PROTECTED] wrote: are you sure the monitor is started and sotoped via the dialplan ? If you're using Monitor() or MixMonitor(), then just add a UserEvent() call just before it in the dialplan. If you're doing monitoring of queues, it's a bit trickier - you have to watch for Join events to see what calls are being enqueued, then when you see a Link event for that call, you can assume (based on local policy) that the monitoring has started (assuming there was no Leave event in the meantime - the logic in your AMI client has to match the logic in your dialplan that deals with queues obviously) If you're talking about automon, there's no support for that, but a cursory examination of the code doesn't show any reason why it couldn't be added. Look at builtin_automonitor() in res_features.c. -- j. ___ --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] Channel name in queue log replaced by a manager event?
Under 1.4.5 and 1.4.6, I've seen a few instances where the channel name in queue log entries is replaced by a snippet of a manager event: --START-- 1183582823|1183582823.104763|queuename|SIP/|REMOVEMEMBER| 1183582828|1183582793.104744|queuename| Context: macro-dialout Extension: s Priority: 3 Application: GotoIf AppData: 0?blockclid Uniqueid: 1183582822.104759 |CONNECT|12|1183582816.104754 1183582833|1183582485.104605|queuename|SIP/|AGENTATTEMPT| ---END--- --START-- 1183762515|1183762515.18034|queuename|SIP/|REMOVEMEMBER| 1183762518|1183762485.18025|queuename| Context: from-somecontext Extension: Priority: 6 Application: Hangup AppData: Uniqueid: 1183762515.18034 |CONNECT|5|1183762513.18033 ---END--- --START-- 1183762659|1183762631.18061|queuename|NONE|EXITWITHTIMEOUT|1 1183762661|1183762485.18025|queuename|: macro-singlequeue Extension: s-TIMEOUT Priority: 1 Application: VoiceMail AppData: XX|uj Uniqueid: 1183762631.18061 |COMPLETEAGENT|5|143|1 1183762665|1183762211.17926|queuename|SIP/|RINGNOANSWER|2 ---END--- Because the text snippet that replaces the channel name always contains newlines, this makes my queue stats program (QueueMetrics) go crazy. The first part of the line is incomplete, and the second is invalid. If the CONNECT event is what gets lost, then the display shows the call as waiting in queue forever. In every case, it's been the channel name that gets replaced, no other field in the queue log lines. I'm assuming it's some pointer badness going on, but I'm not sure what I'll be able to provide beyond this and a basic config snippet to help track it down - I can't reproduce it on demand, and replicating the conditions under which I've observed it requires four PRIs worth of traffic. It does seem that once a given call has it's channel name pointer corrupted, it remains corrupted until the end of the call - I've never seen a ENTERQUEUE with a proper channel name followed by a corrupted CONNECT then a good COMPLETECALLER. So far, the corruption has first appeared when the CONNECT event is logged. Any thoughts, or should I just replicate what I've got into the bugtracker? Thanks -- j. ___ --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] Missing TRANSFER event in queue log when using Local Channels
Has anyone observed a problem where using Local channels with AddQueueMember
results in missing TRANSFER events?
Right now I'm using straight SIP channels when I call AddQueueMember(). I'm
contemplating moving to Local channels because the non-state-based
wrapuptime blows when you have a channel in multiple queues (they can hang
up and get a call immediately so long as it's from a different queue). My
grand plan is to use the 'h' extension in the context where app_queue calls
my agents to invoke PauseQueueMember instead.
The problem is with the /n suffix to the channel name. With it, I lose
TRANSFER events. Without it, the 'h' extension gets invoked as soon as the
call is bridged to the agent.
My agent context looks like this:
[agents]
exten = 491,1,Dial(SIP/491,20)
exten = h,1,PauseQueueMember(|${CUT(CHANNEL,,1)})
When I do something along the lines of:
AddQueueMember(queuename,Local/[EMAIL PROTECTED])
Then as soon as the call is bridged, my 'h' extension gets run:
-- Executing [EMAIL PROTECTED]:1] Dial(Local/[EMAIL PROTECTED],2,
SIP/491)
in new stack
-- Called 491
-- SIP/491-00aa22d0 is ringing
-- Local/[EMAIL PROTECTED],1 is ringing
-- SIP/491-00aa22d0 answered Local/[EMAIL PROTECTED],2
-- Local/[EMAIL PROTECTED],1 answered SIP/427-9d849a90
== Spawn extension (agents, 491, 1) exited non-zero on '
Local/[EMAIL PROTECTED],2'
-- Executing [EMAIL PROTECTED]:1] PauseQueueMember(Local/[EMAIL
PROTECTED],2,
|Local/[EMAIL PROTECTED]) in new stack
-- Stopped music on hold on SIP/427-9d849a90
Once the call is bridged, I transfer to from the agent softphone. This
is what queue log looks like for this type of call:
1183671934|1183671934.5745|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427
1183671937|NONE|NONE|Local/[EMAIL PROTECTED]/n|PAUSEALL|
1183671940|1183671934.5745|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
|CONNECT|6|1183671934.5746
1183672005|1183671934.5745|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
|TRANSFER||from-somecontext|6|65
If I use /n when adding the channel to the queue:
AddQueueMember(queuename,Local/[EMAIL PROTECTED]/n)
Then my 'h' extension is not executed until the bridged call is actually
over. I do the same transfer, but it doesn't show up in the queue log - the
call appears to have been terminated by the caller.
1183672119|1183672119.5839|emerg_nccc_ld_ts|NONE|ENTERQUEUE||427
1183672124|1183672119.5839|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
/n|CONNECT|5|1183672119.5840
1183672135|1183672119.5839|emerg_nccc_ld_ts|Local/[EMAIL PROTECTED]
/n|COMPLETECALLER|5|11|1
Any ideas? I need the Pause-on-agent-hangup behaviour (or something like
it, short of adding proper wrapup state to app_queue), but I can't lose
visibility of my transfers (especially not after I just introduced the sales
people to them after never having visibility of this stat on a Nortel BCM)
Much appreciated.
--
j.
___
--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] TE412 / HPDL380G5 / * 1.4 / CentOS 4.5 Experience
Has anyone successfully run * 1.4 with the following configuration (or something very similar)? HP DL380 G5 (3Ghz Xeon) CentOS 4.5 (kernel 2.6.9-55) Asterisk 1.4.5 (or 1.4.4) Zaptel 1.4.3 (or 1.4.2.1) TE412P TDM400B (2x FXO and 2x FXS modules) I've had this rig running * 1.2.18 with Zaptel 1.2.17.1 for several months without any issues. Upon trying to upgrade to * 1.4.4 and Zaptel 1.4.2.1 a few weeks ago, I saw several kernel panics, easily reproduceable with a load test suite that bridged calls from the PSTN to SIP desk phones. The one time I exposed it to real world traffic (a small office, 30 extensions), I saw five panics in the space of three hours. Interestingly, when the load test only walked through the IVR and queues, I couldn't get it to panic, even if I filled the TE412P and had 90 simultaneous calls going through the system. Only bridged calls seem to cause problems. The panics didn't seem to follow any particular pattern. I saw NMI traps a few times, then failures in the wct4xxx interrupt handler, a few swap tainted errors, etc. To rule out an interaction between the TE412P and the HP motherboard, I moved the installation to an Intel server board (model SE7320) with a 3 Ghz Pentium D. The hard drive was a direct clone from the HP. Running the same load test against 1.4.4/1.4.2.1 - no panics. Running the load test against 1.4.5/1.4.3 - no panics. I'm now nearly 6 hours into having the system exposed to real-world traffic - no panics. At this point, I'm pretty much ready to just put this Intel board into a rackmount server and be done with it, but I am interested to see if anyone else has seen similar problems, or if anyone has run this configuration without any problems. Part of me is saying I can't be the only person to run * 1.4 on a current HP server, which in turn leads me to wonder if this is an incompatibility with the specific board I have, or if I've got a faulty server (though I have run the full HP diagnostic suite several times). Thanks for any feedback you can provide. -- j. ___ --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] Inexpensive Layer 3 Switch?
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote: Any recommendations on an economical layer 3 switch? Preferably something that you have hands on experience with connecting to IP phones with attached PCs? Specifically I need the ability to set the VLAN in the phone to tag voice packets and to set a native VLAN on a per port basis on the switch to put the untagged packets from the attached PC into a separate VLAN. POE is not a requirement but if you have suggestions for an economical layer 3 switch with POE I'd be glad to hear them…so far I'm looking at the SFE2000 from Linksys. I'm using a SFE2000 with PoE with Asterisk. Besides * and my management box (MySQL, ARI, Queuemetrics, etc.), I have the 10 desk phones that I need PoE for plugged into it, a mix of GXP2000, Linksys SPA941 and a couple of Aastra 480i and 9133i units. One of the things that sold me on it was that it can do 185W across all ports; you're not stuck giving 7W or 15W to each port (which was a problem with many models I looked at and limits you to only 12 of 24 ports getting power). I'm told the Grandsteams pull about 4W, and since that's the phone with the widest deployment, I expect to be able to drive 24 per switch eventually. So far I've had no problems with it, though I'm not using it's layer 3 functionality. I trunk two VLANs to a Baystack layer 3 switch, which was pretty simple to set up and has worked properly ever since. My setup doesn't have both tagged and untagged packets coming into the same port, so I can't speak to that. The configuration certainly seems to support it, and I suspect that the default admit all / no ingress filter combined with the fact that every port has to have a PVID assigned means that it would work pretty much out of the box after you configure your VLAN numbers. The UI interface needs some improvement. It's not quite sure if it's a linksys or a cisco right now. You can make all the configuration via a Web GUI as you would with a typical Linksys SOHO router, but if you don't go to Admin - File Management - Copy Files and choose to copy running-config to startup-config (using drop-down boxes, naturally), it loses all your changes on reboot. :) If you have other questions, feel free to contact me off-list. -- j. ___ --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] CNAM.
On 6/17/07, Nick Seraphin [EMAIL PROTECTED] wrote: Yes... 1.5 cents per dip... you prepay the fees... and they deduct from the prepaid amount. You can start with $5.00 which seems like a low-risk to check it out at least. The CLEC I use is more expensive that that for CNAM, and they want to do it on EVERY incoming call, even wrong numbers, whether it's answered or not, per PRI. So since I get several thousand wrong numbers a month, and only 100 or so calls that I actually CARE what the CNAM is on those calls, I can set it up in Asterisk to only do the dip for certain DNIS numbers. I calculated that instead of $70+/month this will cost me $1.50/month. Nice savings. :-) I just hope it's reliable when the call volume picks up more. I gave this a shot yesterday. I figure I can stand to lose $5 if it sucks. Which for someone in Canada, it does. Granted, their website is somewhat hazy on whether or not they support Canadian CNAM - part of the page says can I look up numbers outside the US and Canada while part says outside the US, then the body says we don't support non-NANPA numbers. Pretty much every number I have tried to look up so far for Toronto/GTA just gives me back the city for the name, so I get a bunch of NORTH YORK ON and TORONTO ON or CELLPHONE ON results back, but no actual names. I've gotten a few correct hits back on company numbers, but just as many wrong ones. The Hilton in Edmonton's number comes back as GTCO CALCOMP, and a company I deal with in Mississauga (in the 905 NPA) comes back as ETOBICOKE ON (which is in 416). On the upside, it did find PIZZA PIZZA correctly. /sigh Of course, this is all via their web portal. I am completely unable to connect via their AGI port as provided in their sample configuration page. I get connection refused, which under a stock 1.2 Asterisk drops the call, so I can't leave the dialplan logic intact in the hopes that this is a transient error. Attempts to telnet to the port given via their portal are met with an immediate RST packet, suggesting that their fastagi service is down. At least the cost to play was cheap. IMO, it's not ready for production usage (at least under 1.2 - under 1.4 you can recover from a failure to connect to an AGI service and continue dialplan execution) -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Kernel Panic in wct4xxp during unload on Zaptel-1.4.4
On 5/22/07, Axel Thimm [EMAIL PROTECTED] wrote: Have you tried using the 1.4.x atrpms packages? I did try the 1.4 packages from atrpms overnight yesterday, with similar results. I was able to address the kernel panic when unloading by commenting out ztcfg -s in the stop() function of the init script (based on suggestions on this list). The system appeared stable (went through several clean startup/shutdown cycles), but then proceeded to kernel panic four times in three hours when calls were being processed, forcing me to downgrade to 1.2 again. Unfortunately, I was remote and unable to capture the full kernel panic details, so I'm kind of stuck at square one until I can upgrade again during a maint window and attempt to force a panic during call processing. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kernel Panic in wct4xxp during unload on Zaptel-1.4.4
I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to
1.4 this weekend. Intially everything looked like it was working properly,
but some time in the day following the upgrade, the system died to a kernel
panic. I wasn't able to catch the entire kernel dump on the console
unfortunately.
I attempted to isolate the panic, and found that when 'service zaptel stop'
was run (specifically, when wct4xxp was unloaded) I get this panic
consistently:
5 Not prepped yet! (repeated approx 250 times)
5 Freed a Wildcard
5 Not prepped yet! (repeated approx 550 times)
5 Stopped TE4XXP, Turned off DMA
5 Not prepped yet! (repeated approx 11000 times)
5 Unable to handle kernel paging request at ff034010 RIP:
5 a0163207{:wct4xxp:t4_interrupt_gen2+63}
5 PML4 4ea063 PGD 1388067 PMD 1389067 PTE 0
5 Oops: [1] SMP
5 CPU 1
5 Modules linked in: zttranscode(U) wct4xxp(U) zaptel(U) crc_ccitt
netconsole md5 ipv6 dm_mirror dm_mod button battery ac joydev ehci_hcd
uhci_hcd hw_random bnx2 ext3 jbd
cciss sd_mod scsi_mod
5 Pid: 11053, comm: hotplug Not tainted 2.6.9-42.0.8.ELsmp
5 RIP: 0010:[a0163207]
a0163207{:wct4xxp:t4_interrupt_gen2+63}
5 RSP: :0100013ebdb0 EFLAGS: 00010046
5 RAX: ff034000 RBX: 010073478724 RCX: 0002
5 RDX: 010073478680 RSI: 0002 RDI: 010073478724
5 RBP: 010073478680 R08: 0008 R09:
5 R10: R11: 0002 R12: 00d1
5 R13: 0100013ebec8 R14: 0100013ebec8 R15: 010075e94978
5 FS: 002a955643e0() GS:804e5900()
knlGS:
5 CS: 0010 DS: ES: CR0: 8005003b
5 CR2: ff034010 CR3: 013d8000 CR4: 06e0
5 Process hotplug (pid: 11053, threadinfo 01007413c000, task
01007c445030)
5 Stack: 00d1 01007413dc98 80138552
0038
50100013ebea8 0100013ebde8 0001
00d1
50012 010073478680
5 Call Trace:IRQ 80138552{printk+141}
80112f4a{handle_IRQ_event+41}
5801131c4{do_IRQ+197}
80110833{ret_from_intr+0}
58013c731{__do_softirq+77}
8013c7e5{do_softirq+49}
580110bf5{apic_timer_interrupt+133} EOI
8011c21a{flush_tlb_page+44}
580169106{do_wp_page+1127}
80123ed3{do_page_fault+575}
580169ff2{handle_mm_fault+1228}
80123e9a{do_page_fault+518}
58011026a{system_call+126}
80132bc6{schedule_tail+202}
580110d91{error_exit+0}
5
5 Code: 8b 40 10 89 44 24 58 e8 3d 80 1a e0 31 c0 f6 44 24 58 07 0f
5 RIP a0163207{:wct4xxp:t4_interrupt_gen2+63} RSP
0100013ebdb0
5 CR2: ff034010
5 0Kernel panic - not syncing: Oops
5 Badness in panic at kernel/panic.c:118
5
5 Call Trace:IRQ 80137a8a{panic+527}
80110bf5{apic_timer_interrupt+133}
580111aec{oops_end+38} 80111b07{oops_end+65}
580124148{do_page_fault+1204}
a0078f51{:bnx2:bnx2_start_xmit+470}
5802bb4cd{netpoll_send_skb+257}
80110d91{error_exit+0}
5a0163207{:wct4xxp:t4_interrupt_gen2+63}
80138552{printk+141}
580112f4a{handle_IRQ_event+41}
801131c4{do_IRQ+197}
580110833{ret_from_intr+0}
8013c731{__do_softirq+77}
58013c7e5{do_softirq+49}
80110bf5{apic_timer_interrupt+133}
5 EOI 8011c21a{flush_tlb_page+44}
80169106{do_wp_page+1127}
580123ed3{do_page_fault+575}
80169ff2{handle_mm_fault+1228}
580123e9a{do_page_fault+518}
8011026a{system_call+126}
580132bc6{schedule_tail+202}
80110d91{error_exit+0}
5
5 Badness in i8042_panic_blink at drivers/input/serio/i8042.c:987
5
5 Call Trace:IRQ 8024219b{i8042_panic_blink+238}
80137a38{panic+445}
580110bf5{apic_timer_interrupt+133}
80111aec{oops_end+38}
580111b07{oops_end+65}
80124148{do_page_fault+1204}
5a0078f51{:bnx2:bnx2_start_xmit+470}
802bb4cd{netpoll_send_skb+257}
580110d91{error_exit+0}
a0163207{:wct4xxp:t4_interrupt_gen2+63}
580138552{printk+141}
80112f4a{handle_IRQ_event+41}
5801131c4{do_IRQ+197}
80110833{ret_from_intr+0}
58013c731{__do_softirq+77}
8013c7e5{do_softirq+49}
580110bf5{apic_timer_interrupt+133} EOI
8011c21a{flush_tlb_page+44}
580169106{do_wp_page+1127}
80123ed3{do_page_fault+575}
580169ff2{handle_mm_fault+1228}
80123e9a{do_page_fault+518}
58011026a{system_call+126}
80132bc6{schedule_tail+202}
580110d91{error_exit+0}
5 Badness in
Re: [asterisk-users] `PATH_MAX' undeclared here (not in a function) in asterisk!
On 5/15/07, lizhong zhu [EMAIL PROTECTED] wrote: I compiled asterisk under arm-linux. i am using asterisk 1.4.2. i can run ./configure and menuselect with embedded modules. but running make comes out errors: ranlib libmxml.a [...[ /usr/src/asterisk-1.4.2/include/asterisk/paths.h:23: `PATH_MAX' undeclared here (not in a function) [...] PATH_MAX on the Linux systems I have comes from /usr/include/linux/limits.h, which gets pulled in by a few headers, sys/param.h being the most used one. In my 1.4.4 source tree, this gets pulled in via autoconf, which has this snippet in it's output file include/asterisk/autoconfig.h /* Define to 1 if you have the sys/param.h header file. */ #define HAVE_SYS_PARAM_H 1 Check that you indeed have all your headers installed. If PATH_MAX is in an include file, but not one that gets pulled in by including sys/param.h, then the configure script might need to be updated - best to open a bug and attach your config.log as well as the basic info about your system. I suspect that you're just missing the kernel-headers rpm (or equiv for your Linux flavor). That's where I get my linux/limits.h from: [EMAIL PROTECTED] asterisk-1.4.4]# rpm -q --whatprovides /usr/include/linux/limits.h kernel-headers-2.6.18-8.1.3.el5 [EMAIL PROTECTED] asterisk-1.4.4]# Many default installs do not include the kernel headers - you either have to choose a kernel development package bundle at install time or install them manually after the fact. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log CODECS in CDR's
On 5/11/07, Morgan Gilroy [EMAIL PROTECTED] wrote:
At the moment to find the codecs used I have to look though the sip
trace or show channels/show channel (annoying when you have 50+
channels).
Im just trying to find an easier and quicker way to keep track of the
codecs used to help with debug etc.
The closest variable iv found is, ${SIP_CODEC} Set the SIP codec for a
call
Ill see if NoOp (${SIP_CODEC}) shows the codec that was used without me
setting it though I don't think it will.
Iv looked all over and I cant find anything so it looks like I may have
to hack a ast_set_var into app_dial or chan_sip
1.4 has the CHANNEL function:
pbxlab-01*CLI show function CHANNEL
pbxlab-01*CLI
-= Info about function 'CHANNEL' =-
[Syntax]
CHANNEL(item)
[Synopsis]
Gets/sets various pieces of information about the channel.
[Description]
Gets/set various pieces of information about the channel.
Standard items (provided by all channel technologies) are:
R/O audioreadformatformat currently being read
R/O audionativeformat format used natively for audio
R/O audiowriteformat format currently being written
R/W callgroup call groups for call pickup
R/O channeltypetechnology used for channel
R/W language language for sounds played
R/W musicclass class (from musiconhold.conf) for hold music
R/W rxgain set rxgain level on channel drivers that support
it
R/O state state for channel
R/W tonezone zone for indications played
R/W txgain set txgain level on channel drivers that support
it
R/O videonativeformat format used natively for video
When I put this in a dialplan with NoOps and called channel macros, I can
kind of get what you're describing:
[from-external-pbxtel]
exten = 491,1,NoOp(${CHANNEL(audioreadformat)})
exten = 491,n,NoOp(${CHANNEL(audiowriteformat)})
exten = 491,n,NoOp(${CHANNEL(audionativeformat)})
exten = 491,n,Dial(SIP/491,20,M(logcodec))
exten = 491,n,Hangup
[macro-logcodec]
exten = s,1,NoOp(${CHANNEL(audioreadformat)})
exten = s,n,NoOp(${CHANNEL(audiowriteformat)})
exten = s,n,NoOp(${CHANNEL(audionativeformat)})
Console output is:
-- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(IAX2/pbxtel-01-5,
ulaw) in new stack
-- Executing [EMAIL PROTECTED]:4] Dial(IAX2/pbxtel-01-5,
SIP/491|20|M(logcodec)) in new stack
-- Called 491
-- SIP/491-0a16d1c0 is ringing
-- SIP/491-0a16d1c0 answered IAX2/pbxtel-01-5
-- Executing [EMAIL PROTECTED]:1] NoOp(SIP/491-0a16d1c0, slin) in
new stack
-- Executing [EMAIL PROTECTED]:2] NoOp(SIP/491-0a16d1c0, slin) in
new stack
-- Executing [EMAIL PROTECTED]:3] NoOp(SIP/491-0a16d1c0, gsm) in new
stack
== Spawn extension (from-external-pbxtel, 491, 4) exited non-zero on
'IAX2/pbxtel-01-5'
-- Hungup 'IAX2/pbxtel-01-5'
This is a call coming in as ulaw over IAX2, then going to a SIP softphone
configured for only gsm.
Hope that helps.
--
j.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sound files
On 5/8/07, Pedro Silva [EMAIL PROTECTED] wrote: Can i identify the sound files that are played in the asterisk console? I defined the verbose to 100 but i can not see the sound files that are played in some situations... :( For example, I need to know what files are played for the message: Extension xxx is unavailable The goal is to translate that to Portuguese (pt_pt)... Try this page: http://www.nathanpralle.com/software/ast_masterlist.html Not 100% up to date, but it covers most of the prompts I'd had to look up. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Display Caller ID of called party
On 5/1/07, Savoy, Kevin - Williston, ND [EMAIL PROTECTED] wrote: Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in case that matters. On a Grandstream GXP-2000, this happens when the dialed number is in the XML phonebook that the phone sucks down from my provisioning server. It might work the same on the Polycom units. Of course, you need to have some process in place to keep the phonebook file up to date. To do it in a generic way where the name is looked up by * and sent back to your phone for display as part of the 100 Trying or 180 Ringing responses, is an entirely different matter. I suspect that the end-user experience would vary wildly based on the equipment each user was using. If this is possible, I'm sure people more knowledgeable than me will chirp in. The phonebook route might be the quickest bang for your buck though. If I recall from testing Polycom phones, you can have a central phonebook shared by all phones and a per-phone phonebook that is uploaded by the phone to your TFTP server so that even when re-provisioning from factory reset, nothing is lost. I didn't get far enough in the evaluation to set up a provisionin server of my own. The evaluation died in committee when an exec reported that she didn't like the small buttons on the IP430. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
On 4/25/07, Mike Lynchfield [EMAIL PROTECTED] wrote: may i add , eyebeams confnig file is xml and could be generated , BUT, the password is hashed in some way.. any idea on that ? its a pretty long hash Two options: - type in the passwords manually, shut down eyeBeam, and then cut/paste the hash into your provisioning system (suggested earlier in this thread) - say screw it and put the password in the XML in the clear and change the attribute on the encrypted attribute from true to false. That just pushes the security concerns to the provisioning system and the workstation filesystem, but you have much more flexibility in how you tackle that problem than you get with the softphone itself. Server side isn't really any worse than most desk phones that need a config file with the password in the clear on a TFTP server (i.e. Grandstream). Client side you have to have proper user-only creds on the Documents and Settings folder for each user and have some way to ensure that your users don't give access to the filesystem as them to someone else (i.e. forgetting to lock the screen). But that's a PEBKAC[1] issue best addressed through the administration of a Big Foam Cluebat(tm). [1] Problem exists between keyboard and chair -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
On 4/20/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: I am not. The soft phone is not the only software on that computer that needs cetral configuration. How do you configure the networking on those computers? The mail clients? How do you deploy updates? The fundamental problem, as I explained in my first response, is that X-Lite/eyeBeam will not search for it's configuration in the part of a user's Documents and Settings folder that is part of a roaming profile. I allow that I may just be a complete Windows admin neophyte, but my understanding is that the Documents and Settings\username\Local Settings folder is machine-specific. X-Lite and eyeBeam store their configurations in this folder, and it is not copied back to the server on logout nor pulled from the server when the user logs into a different workstation. It was my hope that it might look in the Local Settings folder, and if the appropriate directory was not found then continue the search in the Documents and Settings\username\Application Data folder, but this does not appear to be the case. So, it is more accurate to say that I want a softphone that I can configure in an automated fashion (preferably via text or XML configs) from a central server. Whether that is part of a Windows user profile or whether it is pulled via TFTP/FTP/HTTP is irrelevant to this discussion. I merely stated it because it is what SIP desk phones tend to support and I am trying to build something analogous with a softphone without reinventing the wheel. -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Softphone that supports central provisioning?
On 4/21/07, Senad Jordanovic [EMAIL PROTECTED] wrote: What about creating a configuration file on server for each soft phone extension automatically and then importing that file into the soft phone? In another words, user receives a link to the setup program and the configuration file in an email. That is how we are configuring our soft phones :-) This is essentially what I have done. I've configured X-Lite for user #1, then shut it down and copied the config folder to shared location named for their windows logon. Then I start up X-Lite, reconfigure for user #2, shut it down and copy the config folder to a shared location named for their windows logon. The logon batch file looks for configs in the shared location and pulls them to the local PC, overwriting the existing configuration. This also provides an escape hatch should the user bork the configuration of X-Lite - just log out and back on again. Workable, but inelegant. And not without other problems - I have a mix of different headsets in the office, and when an X-Lite configuration made with one type of headset starts up on a PC with another type of headset, it tends to reset it's configuration to point the headset speaker to the PC's speakers. Quite annoying, and made even more so by the aforementioned escape hatch. As I've mentioned, I'm just looking for a better way of doing this without reinventing the wheel. Along a timeline of the appropriate length, I can make any of the suggested solutions work. Anyone selling a timeline lengthener? :) -- j. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
