Re: [asterisk-users] I think this is a bug (video call file) 11.25.1 and 13.13.1
On 12/20/2016 06:01 PM, Jerry Geis wrote: >Hi Jerry, > just had a look through the code, and from what I can tell, what >you're trying to do is not supposed to work, exactly. It appears that >what Asterisk expects is to be given a filename, such as "myplayback". >Asterisk will first search for an audio version of the file (like >myplayback.gsm or myplayback.opus), and open that as an audio stream. If >that succeeds, it then will also see if there is an accompanying video >stream (such as myplayback.h264). If it then finds that video, then the >result will be that Asterisk will play the audio from the audio file and >the video from the video file. >What this means is that Asterisk does not properly handle: >* Files that have audio and video streams contained within >* Video files without accompanying audio >This is one of those times where Asterisk's handling of video is not >user-friendly and in general ass-backwards and terrible. If you have a >tool that can extract the audio to its own file, then you would be able >to run your scenario, presumably. >It would be a welcome addition for Asterisk to be able to open a single >file containing video and accompanying audio and be able to play those back. Hi Mark, Thanks for your reply... I just tried what you suggested on only got audio. I created a wav file and put it in the /tmp directory just like the video.h264 file. So /tmp has video.h264 and video.wav both. I then placed the call and only heard the audio from the wav file. I used this for my call file: Channel: SIP/2002 Context: testing Extension: 99 Priority: 1 Application: Playback Codecs: h263,h264,vp8,g722,ulaw,alaw,wav Data: /tmp/video My Bria 4 softphone uses the h263 and h264 codecs and of course wav file audio. Based on your look of the code did I miss something to trigger the playing of the video file? I can extract the audio out to a seperate file - so not a show stopper for me. No errors showed up on the Asterisk CLI when I did my test. Thanks so much, Jerry I don't see anything obvious in the code that would have prevented the video from playing back. Unfortunately, the debug from Asterisk isn't going to be especially helpful here, with one exception. If you have core debug at level 1 or higher, then when Asterisk detects the video file, it will say: "Ooh, found a video stream, too, format h264" If you see that message, that at least means that Asterisk is finding the video file as expected. If you don't see that, then it's likely that Asterisk is unaware of the h264 file format type. It may be that you don't have the format_h264.so module loaded. It may be that there was an error that occurred when that module was loading, causing it not to be able to load properly. If you are seeing that debug message, it at least means that Asterisk attempted to play back the video file, but something else in the process caused the video not to play back as expected. The first thing you could check is packet captures to see if Asterisk is even attempting to send video to the softphone. If so, then it's likely that there is some sort of codec mismatch happening (likely something in the format parameters). If Asterisk is not even attempting to send any video, then it likely means that there is some other issue. It may be a bug, or it may be some erroneous condition in the environment. Hard to tell yet though. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I think this is a bug (video call file) 11.25.1 and 13.13.1
On 12/19/2016 07:39 PM, Jerry Geis wrote:
I can create an audio call file and specify Application: Playback and Data: a
path to the audio file, it calls the phone and plays the audio message just
fine.
I am trying to do the same with a video file. I specify Application: Playback
and Data: the path to the video file (no ending of course), and I do specify
also the Codecs: h264,h263 etc...
Asterisk reports:
*File /tmp/video does not exist in any format
*>* Unable to open /tmp/video (format ulaw|h263|h264)*
Looking then at the code and attaching with the debugger.
the ast_openstream_full() function has this condition:
if (!fileexists_core(filename, NULL, preflang, buf, buflen,
file_fmt_cap) || !ast_format_cap_has_type(file_fmt_cap, AST_MEDIA_TYPE_AUDIO)) {
}
So fileexists_core() returns 1 but the next call to ast_format_cap_has_type()
fails. because its looking for AST_MEDIA_TYPE_AUDIO and the file is a video
file. Nowhere is the an AST_MEDIA_TYPE_VIDEO.
I can use the call file to setup a video call between two video softphones just
fine. However using the call file to call a phone and play a video is not
working at all for me.
Am I on the right track?
Is this supposed to work? if so how since there is no check of the
AST_MEDIA_TYPE_VIDEO?
Thanks,
Jerry
Hi Jerry,
I just had a look through the code, and from what I can tell, what
you're trying to do is not supposed to work, exactly. It appears that
what Asterisk expects is to be given a filename, such as "myplayback".
Asterisk will first search for an audio version of the file (like
myplayback.gsm or myplayback.opus), and open that as an audio stream. If
that succeeds, it then will also see if there is an accompanying video
stream (such as myplayback.h264). If it then finds that video, then the
result will be that Asterisk will play the audio from the audio file and
the video from the video file.
What this means is that Asterisk does not properly handle:
* Files that have audio and video streams contained within
* Video files without accompanying audio
This is one of those times where Asterisk's handling of video is not
user-friendly and in general ass-backwards and terrible. If you have a
tool that can extract the audio to its own file, then you would be able
to run your scenario, presumably.
It would be a welcome addition for Asterisk to be able to open a single
file containing video and accompanying audio and be able to play those back.
Mark Michelson
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Re: [asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header?
On 07/10/2015 11:53 AM, Rodrigo Pimenta Carvalho wrote: Hi. The ASTERISK wiki has a page showing the function PJSIP_HEADER(). However, it doesn't explain if such function works only over SIP INVITE messages or if it can be use, for example, to read headers from others types of SIP messages too. So, can I use PJSIP_HEADER to read the SIP 183 message header? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) Unfortunately, PJSIP_HEADER() cannot be used on responses because SIP responses do not enter the dialplan. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PJSIP Authrentication by IP fails
On 09/24/2013 02:15 AM, CDR wrote: I have one single endpoint [inhouse](endpoint-basic) type=endpoint and one section like this [indentify] endpoint=inhouse match=X.Z.Y.X You are missing type=identify from this section. The common model of configuration sections in pjsip.conf is that you can name them whatever you wish. The important piece of information when determining what type of configuration section it is is the type= option for the section. With no type= option set, the configuration section is completely ignored. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Case-sensitivity of Dialplan variables.
Hi!
I've been confronted with an interesting issue to resolve. The
issue is located here:
https://issues.asterisk.org/jira/browse/ASTERISK-20163
The issue involves case-sensitivity of channel and global variables
in the dialplan. Current behavior is as follows:
1) Variables created in the dialplan by users are case-insensitive.
Thus if the variable MARK were set, then ${MARK} and ${mark} would
both evaluate to the set value.
2) Variables used internally by Asterisk are case sensitive. So if
some application set a variable called MARK, it would be different
from a variable set by some application called mark.
First off, this inconsistency is just weird. It would be much easier
to just have things work one way or the other, not to have this mix.
In addition, this can lead to some awkward situations. Consider that
someone wants to use a specific SIP codec and so they set the
variable SIP_CODEc to be g722. Notice that the final 'c' is
lowercase, presumably due to a typing error. The option would not
take effect because chan_sip specifically checks the value of the
case-sensitive ${SIP_CODEC}. What makes this weirder is that if
the dialplan writer were to check ${SIP_CODEC} in the dialplan
using a NoOp or Verbose call, then he would see the variable set
to the value he set it to when he set ${SIP_CODEc} because the
variable substitution is case-insensitive in the dialplan. This
makes debugging the problem difficult.
I propose that dialplan variables need to be made consistent in their
evaluation. We need to choose either to be always case-sensitive or
always case-insensitive. The problem is, I don't know which of these
changes would have a larger effect on people. This is where I would
like your feedback. Which way should it go?
Some of you might be eager to propose a configuration option to
decide which it should be. I'm sick of having hundreds of options
in Asterisk to slightly tweak the behavior one way or another. This
needs to go one way or the other, not be configurable.
What I plan to do, no matter which way the vote goes, is to document
on the wiki how things currently behave in Asterisk, to include the
example I gave above (or something similar anyway). Depending how
the vote goes, I will make the necessary code changes in Asterisk
trunk. I will document the behavior change both in UPGRADE.txt and
on the wiki.
When considering which way you lean, consider that we really don't
have much of a precedent to go on. For instance, dialplan applications
are case-insensitive (answer and Answer and ANSWER are all the
same). Dialplan functions, on the other hand, are case sensitive
(HASH would be evaluated properly but hash would not). My personal
opinion is that all variable evaluations should be case-sensitive.
I don't feel all that strongly about it though and could easily be
swayed the other way if people respond overwhelmingly in opposition.
So respond here and let me know what you think. I got a couple of
replies on the -dev list and they said that this would be good to put
out on the -users list too.
Mark Michelson
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Re: [asterisk-users] Pickup calls coming from queues
On 04/16/2012 05:21 AM, Niccolò Belli wrote: I suspected it, but it didn't work at first. I fear I didn't understand what the context refers to in Pickup(extension[@context]). I will make an example: phone-100 wants to pick up a ringing phone-200 (call comes from my-sip-provider). This is my sip.conf [phone-100] context=context-100 [phone-200] context=context-200 [my-sip-provider] context=from-my-sip-provider This is my extensions.conf [context-100] exten = test,hint,Queue:MyQueue exten = test,1,Pickup(myphonenumber@from-my-sip-provider) [...] [context-200] [...] [from-my-sip-provider] exten = myphonenumber,1,Queue(MyQueue,r) same = n,Hangup() I expected to use from-my-sip-provider as context in Pickup, unfortunately it didn't work. So I tried both context-100 and context-200 as context in Pickup and they *both* worked! What's the logic behind Pickup's context? Thanks, Niccolò There is some missing information here. What is the strategy of the queue? How are the queue members listed (i.e. are they SIP channels or local channels)? My suspicion is that the queue is simultaneously dialing local channels in contexts [context-100] and [context-200]. Since there are no ringing channels in context [from-my-sip-provider] there are no calls to pick up there. However, since [context-100] and [context-200] both have ringing channels, doing a call pickup in either of these results in a successful pickup. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RPID on called party
Ondrej Valousek wrote: Hello, Did anyone manage to force asterisk to put Remote-party-ID attribute on the SIP outgoing call? I.e. When A calls B, I want that A gets a name of B displayed on his phone. Note that name of A gets displayed on the B's phone fine, but this is not what I want. This works with Cisco Call manager fine - the RPID is sent as a part of the response to the SIP INVITE this way: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.60.20:5060;branch=z9hG4bK42892c32;rport From: Ondrej Valousek sip:7...@192.168.60.20 sip:7...@192.168.60.20 ;tag=as4786d518 To: sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;tag=f75ff5d8-1023-4240-bc4b-d7eeb6d0d77d-42063104 Date: Tue, 30 Mar 2010 13:53:15 GMT Call-ID: 465a9c200587260d164f451409489...@192.168.60.20 CSeq: 102 INVITE Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY Allow-Events: presence *Remote-Party-ID: Paul Ryan sip:1...@192.168.62.12 sip:1...@192.168.62.12 ;party=called;screen=yes;privacy=off* Contact: sip:1...@192.168.62.12:5060 sip:1...@192.168.62.12:5060 Content-Length: 0 But I can not make it working with Asterisk. Does anyone have any glue how to achieve this WITHOUT patching asterisk? I am happy to upgrade to the latest/greatest version, I just do not want to patch. Many thanks, Ondrej This feature is in Asterisk trunk and will be present in the upcoming 1.8 release. By setting sendrpid=yes on A's phone, Asterisk will send a Remote-Party-ID header that corresponds to what Asterisk received from B. Also, there is a CONNECTEDLINE() dialplan function that can be used to send this information prior to a call. I actually gave a presentation on this topic at Astricon last year, but for some reason the Astricon '09 archive does not seem to have my presentation video available. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to run Music while looking for the caller in Database
Bharath B. Reddy Bynagari wrote: Hi, We are using Asterisk and PERL. We have all the call logic in PERL. We are trying to identify the caller using the CID in the Database. As the Database lookup is taking more time (15 seconds), we want to play some tune while the caller is waiting. How can we do that? Any ideas will be greatly appreciated. - Bharath Assuming that you are using a perl AGI script, you can use the AGI command SET MUSIC ON to play music on the channel. You can stop the music by using SET MUSIC OFF. For a bit more information, you can run the CLI command agi show set music Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reset personal voicemail settings
Felix Tiefenthaler wrote: Hi list, can anyone tell me how to reset/delete all modifications (personal greeting message, personal name, ...) I made in my voicemail? I just want to get the default automatic computer messages back. thank you! greets felix If you are storing voicemail on the file system, then you can just go to /var/spool/asterisk/voicemail/context/mailbox/ and delete the items in there that you want to. The INBOX and Old folders contain new and old messages. Anything else in there will be greetings and other similar recordings. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp.conf ports for inbound or outbound?
Michelle Dupuis wrote: I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? Thanks! MD The port range specified is used for both inbound and outbound calls. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp.conf ports for inbound or outbound?
Olle E. Johansson wrote: 25 mar 2010 kl. 13.14 skrev Michelle Dupuis: I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? These are the ports Asterisk use for INCOMING media, the ports we listen on. /O Yes, I may have misinterpreted the question in my previous response. Asterisk will use this port range on both incoming and outgoing calls. However, the port range specified, as Olle says, is the range used in SDPs and thus the port that Asterisk will instruct the far end to send the media to. Mark Michelson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6.0.11-rc2, 1.6.1.2, 1.6.1.3-rc1, and 1.6.2.0-beta4 Release Announcement
Alex Hermann wrote: On Monday 03 August 2009, Asterisk Team wrote: The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The related security advisory AST-2009-004 has been released along with this announcement. Please read that advisory for more information. For a full list of changes in these releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.6 .1.2 The chaneglog doesn't mention anything on fixing a security issue. Even worse, the changelog doesn't mention anyting at all besides the version increment. Is the fix really applied? The fix is applied. I just checked to be sure. I can't say for sure why the change did not show up in the changelog, but I'm guessing the reason is that the tag for the release was created first, and then the specific fix was applied to the tag instead of creating the tag based off an already-fixed branch. This was an oversight on our part, and we'll do our best not to make such a mistake again. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail feature: enable or disable the ability to leave a message
Emrah wrote: Hi, I think there is an essential option of the Voicemail application that is missing. I would like to suggest the implementation of a function to give the user the ability to either allow or disallow the recording of messages. If the ability to record a message is disabled, options u, s, and b must not be considered in order to avoid the playback of messages such as Please leave your message after the tone... the usecase is simple. A person could record a greeting that says please callback later instead of asking to leave a message. usefull also to record afterhour messages. What do you think? Regards, Emrah There is no reason to place this logic in the Voicemail application itself. If you wish to give users the option of leaving a voicemail, it can easily be done in the dialplan by playing prompts and reading the input of the user. Then, based on the input, you can choose whether to run the Voicemail application. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Updated patch for 8824?
Benny Amorsen wrote: Doug Lytle supp...@drdos.info writes: I don't suppose anybody has an updated CalledID patch for Asterisk 1.4.26: https://issues.asterisk.org/view.php?id=8824 I've been running 1.4.21.1 for a while and have the need to update one of our Asterisk Conference/Faxserver/switches and wanted to see if the patch available would apply. It failed in several places. This keeps me from updating my other installs. You could try 1.6.x :) (It's apparently available in all 1.6.x branches, but I'm not sure whether it made it into actual releases yet). /Benny Actually, this feature is only in Asterisk trunk currently. It will be present in 1.6.3 once it has been branched. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Mark Michelson wrote:
You need to set a call-limit for the SIP peer. Device state calculation for
a
SIP peer is predicated on both the call-limit and busylevel. Let's say that
you
were to have a call-limit of 2, but no busylevel set. These are the device
states reported for the peer based on the number of calls currently handled:
Hi Mark. Thanks for your explanation of these parameters.
I should have posted my configurations. I double-checked the contents
of sip.conf and I have this. The 'subscribecontext' was added for
testing, per the other reply I got for my question.
;
; Settings common to all devices on our system
;
[basic-options](!)
type=friend
host=dynamic
canreinvite=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
qualify=yes
;
; Standard desksets here
;
[lan-deskset](!,basic-options)
context=sip-deskset
notifyringing = yes
notifyhold = yes
limitonpeers = yes
call-limit=99
[6668](lan-deskset)
secret=mysecret
callerid=Matts SIP 6668
username=Barry's IP450
call-limit=32
busylevel=1
subscribecontext=hint-context
My hint-context is:
[hint-context]
exten = 6668,hint,SIP/6668;
I'm still not getting anything other than NOT_INUSE from DEVICE_STATE.
Here is the CLI output:
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:1]
NoOp(SIP/-0955ecc8, SIP/6668 has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:2]
NoOp(SIP/-0955ecc8, SIP/ has state NOT_INUSE) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:3]
ExecIf(SIP/-0955ecc8, 0?Busy(10)) in new stack
[Jul 15 18:40:15] -- Executing [6...@sip-deskset:4]
Dial(SIP/-0955ecc8, SIP/6668) in new stack
And here is sip show inuse:
corp-asterisk*CLI sip show inuse
* User name In use Limit
6668 1 32
6667 0 99
1 99
* Peer name In use Limit
6668 1/1/0 32
6667 0/0/0 99
0/0/0 99
For completeness, here is the dialplan that's producing this:
exten = 6668,1,NoOp(SIP/${EXTEN} has state ${DEVICE_STATE(SIP/${EXTEN})});
exten = 6668,n,NoOp(SIP/ has state ${DEVICE_STATE(SIP/)});
exten = 6668,n,ExecIf($[${DEVICE_STATE(SIP/${EXTEN})}=BUSY]?Busy(10));
exten = 6668,n,Dial(SIP/${EXTEN});
Thanks for the config info. I have a couple of suggestions for fixes.
1. Try changing the type in [basic-options] from friend to peer. I've found
that
device state reporting for outbound calls (from the perspective of the phone)
tends to be more accurate with this type.
2. If for some odd reason number 1 either doesn't sound appealing to you or
doesn't work, then try moving the limitonpeers=yes option from your
[basic-options] section to the [general] section.
No, neither of these ideas actually make any real sense to me, but they are
based on behavior that I have witnessed with my Asterisk setup in my office.
Mark Michelson
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Re: [asterisk-users] DEVICE_STATE() and Asterisk 1.6.0.10
Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than NOT_INUSE. I have two extensions: and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use to call 6668 and in the dialplan have a noop to see what DEVICE_STATE() is returning for both extensions. I get: [Jul 15 17:20:43] -- Executing [6...@sip-deskset:1] NoOp(SIP/-08636430, SIP/6668 has state NOT_INUSE) in new stack [Jul 15 17:20:43] -- Executing [6...@sip-deskset:2] NoOp(SIP/-08636430, SIP/ has state NOT_INUSE) in new stack 6668 is configure so that I get this: * Name : 6668 Secret : Set MD5Secret: Not set Context : sip-deskset Subscr.Cont. : Not set Language : AMA flags: Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 99 Busy level : 1 Dynamic : Yes Callerid : Matts SIP 6668 MaxCallBR: 384 kbps Expire : 2016 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID: No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : Addr-IP : 192.168.1.70 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Transport: UDP Def. Username: Barry's IP450 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw:20) Auto-Framing : No 100 on REG : No Status : OK (14 ms) Useragent: PolycomSoundPointIP-SPIP_450-UA/3.1.3.0439 Reg. Contact : sip:6...@192.168.1.70 Qualify Freq : 6 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs - From what I have read, with 'Busy Level = 1' I should be seeing BUSY returned from the DEVICE_STATE() call, yet I don't. What is the super-secret sauce required to get Asterisk to return the correct state? TIA, Barry You need to set a call-limit for the SIP peer. Device state calculation for a SIP peer is predicated on both the call-limit and busylevel. Let's say that you were to have a call-limit of 2, but no busylevel set. These are the device states reported for the peer based on the number of calls currently handled: 0 calls: not in use 1 call: in use 2 calls: busy Basically, the busylevel defaults to the call-limit value. Now if you add a busylevel = 1 to sip.conf, these are the device states reported: 0 calls: not in use 1 call: busy 2 calls: busy Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No response to our critical packet problem
James Lamanna wrote: Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with no response to our critical packet. Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect calls to internal office extensions (which still go through asterisk) OR voicemail 2) The other 20+ phones in the same office on the same network have 0 problems. Here's a SIP trace of the problem. yyy.yyy.yyy.yyy is the outside NAT IP xxx.xxx.xxx.xxx is the IP of my PBX dd is the dialed phone number sss is the source phone number The peculiar thing is that asterisk sends an OK in response to an INVITE, then the phone sends back an ACK, which asterisk seems to ignore because it retransmits the OK message again Then eventually the phone gives up and sends a BYE message. -- James I think I know what the problem is here. It's not the fault of the phone, but of Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#', specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq in a dialog, so once the INFO arrives, we no longer have any memory of the Cseq of the INVITE that the phone sent. Later, we send a 200 OK response for the INVITE. Then, when we receive the ACK from the phone, we drop it since it's Cseq is less than the latest Cseq we received in this dialog. As a result, Asterisk never realizes that it has received the ACK. Asterisk continues retransmitting a 200 OK to the phone and the phone dutifully keeps sending an ACK in response until Asterisk has retransmitted the maximum amount of times. There are a couple of potential ways of solving this issue. One is to add an Answer to your dialplan as the first priority. This way, the INVITE is completely answered before the phone ever sends any INFO requests. Another is to switch the phone away from using INFO to transmit DTMF. I would be willing to bet that the other phones on your network are not using INFO for transmission of DTMF, and so they are not experiencing the same issue. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No response to our critical packet problem
Martin wrote: I think I know what the problem is here. It's not the fault of the phone, but of Asterisk. The phone is sending an INVITE and then an INFO (DTMF '#', specifically) to Asterisk. Asterisk only keeps track of the last incoming Cseq in a dialog, so once the INFO arrives, we no longer have any memory of the Cseq of the INVITE that the phone sent. well then Asterisk now behaves as a poor written hand script that handles SIP calls ... INFO can arrive at any time when dtmfmode=info Martin Yes, this would be why I said that it is Asterisk's fault and provided possible workarounds. Thank you for your helpful and constructive criticism. Mark Michelson ___ -- 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] interruption in queue
Kevin P. Fleming wrote: Rilawich Ango wrote: I want to allow user to press 0 to the voicemail if the user don't want to wait in the queue. Below is what I set but it doesn't work. Anyone can help? ango None of that is necessary, but reading the documentation is. app_queue already supports the caller using a DTMF key to exit from waiting for an agent; 'core show application queue' should give you the information you need. Not to undermine Kevin's requests to read what is documented, I can say that what you want actually will not be presented by running core show application Queue in the CLI. What you need to look at is the queues.conf.sample file in the configs/ directory of the source. There you will find a myriad of options you may set for a queue, including the DTMF exit option you want. Mark Michelson ___ -- 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] Spiral SIP Request problem
amit salunkhe wrote: Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the extension of the party you wish to reach etc !! 2) Whenever a user presses any 4 digit number, asterisk routes that call back to Openser to find if the user is available there. 3) In Opensips the preference for the user is set such that all call are supposed to go to voicemail which is handled by the same asterisk again; so a 0 is prefixed to the R-URI and sent back to asterisk by opensips. 0 is prefixed so that asterisk does not generate loop detected for the INVITE. 4) The call gets answered by asterisk but the 200 OK message keeps routing between asterisk and opensips and then asterisk times out with “*Maximum retries exceeded on transmission*” error. Scenario: 1) User 7...@sipproxy.comà Opensips 7...@asterisk.com-àasterisk 2) Asterisk --200ok-à Opensips---200ok---àUser 3) Asterisk waits for extension input and user presses 7010 4) Asterisk--INVITE 7...@sipproxy.com-Opensips https://remote.novanet.net/owa/redir.aspx?C=31661696fae74c3a94f26b78ea106eaeURL=mailto%3a7010%40sipproxy.com-%253eOpensips 5) Opensips checks the preference for user 7010 and needs to send this call to voicemail of 7010 which is handled again in asterisk. 6) Opensips-INVITE 07...@asterisk.com--Asterisk https://remote.novanet.net/owa/redir.aspx?C=31661696fae74c3a94f26b78ea106eaeURL=mailto%3a07010%40asterisk.com--%253eAsterisk (0 is added in the R-URI in order to avoid loop detected by asterisk) 7) Asterisk plays minivm greet and gets ready for minivm record although no audio is heard by the user 8) Asterisk200ok-àOpensips (Since the call to voicemail by Opensips is answered) 9) Opensips---200ok-àAsterisk (Since the call to 7010 by asterisk is answered) 10) Asterisk does not ACK the 200ok in step 9 and instead keeps sending the 200ok in step 8 until maximum retransmission is reached and then the call is hung up. w/regards, Amit What Asterisk version are you using? I'm guessing that since you are using minivm, you are likely using a 1.6 version of Asterisk. I recently (as in earlier this week) committed some code to the 1.6 branches to improve the spiral support in Asterisk. If you are able to retry with the latest subversion checkout of the branch you are using, you may find that things are working better now. Mark Michelson ___ -- 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] meetme dies looking for conf-getconfno
sean darcy wrote: Danny Nicholas wrote: You lost conf-getconfno.gsm . Asterisk is trying to play that file to let you pick a conference number to use. It goes in /var/lib/asterisk/sounds. Grep for it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, May 15, 2009 12:39 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] meetme dies looking for conf-getconfno With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. cat meetme.conf [rooms] conf = 600 extensions.conf: [meetme] exten = 2663,1,MeetMe(,D) exten = 2663,n,Hangup() exten = 2666,1,MeetMe() exten = 2666,n,Hangup() What I'm expecting is to dial 2663, get a conference room number ( 600, I suppose since it's the only room ), and set a PIN. Then I'd dial 2666 enter the conference room number and the PIN, and be put in conference. But here's what happens when I dial 2663: -- Starting simple switch on 'DAHDI/1-1' -- Executing [2...@internal:1] MeetMe(DAHDI/1-1, ,D) in new stack [2009-05-15 13:21:19] WARNING[2061]: file.c:641 ast_openstream_full: File conf-getconfno does not exist in any format [2009-05-15 13:21:19] WARNING[2061]: file.c:924 ast_streamfile: Unable to open conf-getconfno (format 0x4 (ulaw)): No such file or directory == Spawn extension (internal, 2663, 1) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' conf-getconfno does exist: ls -l /var/lib/asterisk/sounds/en/conf-getconf* -rw-r--r--. 1 root root 25211 2009-03-26 14:42 /var/lib/asterisk/sounds/en/conf-getconfno.ulaw -rw-r--r--. 1 root root 50466 2009-03-26 14:42 /var/lib/asterisk/sounds/en/conf-getconfno.wav Any help appreciated. sean Will do, but why gsm? Nobody's using gsm, and it looks like it's seeking ulaw ( which is installed). sean You may want to try setting languageprefix=yes in asterisk.conf in the [options] section. If that does not work, then you may wish to try to move the file up one directory level and see if it plays, then. Mark Michelson ___ -- 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] PauseMonitor() Hanging Up Call
Jon Morgan wrote:
Hi All,
I’m at the end of my tether here and would really appreciate some help.
I’m trying to implement DTMF based pause/resume of call recording. I’m
using Asterisk 1.4.22.1.
Here’s the scenario:
The caller (SIP or ISDN, doesn’t matter) dials into the asterisk which
executes the following code:
exten = _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)
exten = _X.,n,Set(__DYNAMIC_FEATURES=in-pauseMonitor#in-resumeMonitor)
exten = _X.,n,Dial(SIP/myphone,300,tTo)
My [applicationmap] in features.conf is setup as follows:
in-pauseMonitor = *7,self/callee,Macro,pause-record
in-resumeMonitor = *9,self/callee,Macro,resume-record
I also have the following contexts setup in extensions.conf:
[macro-pause-record]
exten = s,1,Playback(sounds/recPaused)
exten = s,n,PauseMonitor()
exten = s,n,MacroExit
[macro-resume-record]
exten = s,1,Playback(sounds/recResumed)
exten = s,n,UnPauseMonitor()
exten = s,n,MacroExit
Now, if I setup the call and hit *7 on the callee phone, the call is
hungup every time! No error message, just simply hangs up, as follows:
Executing [...@macro-pause-record:2]
PauseMonitor(SIP/myphone-09d26e60, ) in new stack
== Spawn extension (macro-pause-record, s, 2) exited non-zero on
'SIP/myphone-09d26e60' in macro 'pause-record'
== Auto fallthrough, channel 'SIP/xlite-09d18fc0' status is 'ANSWER'
If I change the [applicationmap] entries in features.conf to allow
pause/resume from the caller phone, e.g.:
in-pauseMonitor = *7,self/*caller*,Macro,pause-record
in-resumeMonitor = *9,self/*caller*,Macro,resume-record
Then it works like a charm!
Seems there’s an issue with pause/resume from callee side.
Can anyone shed any light on what I’m doing wrong here please?
Regards,
Jon Morgan.
The problem is that the callee's channel does not have a monitor on it, just
the
caller's channel. The PauseMonitor application has the unfortunate effect that
if the channel on which it is called has no monitor attached, then the
application returns as if an error occurred and the dialplan stops. I
unfortunately do not see a direct way to tell from the dialplan if a channel
has
a monitor attached (there is a MONITORED channel variable, but it will be true
for both channels of a monitored call).
I can think of ways to work around the problem of the call being hung up, but
the problem is that even with the workarounds in place, calling PauseMonitor on
the callee's channel will not result in the monitor actually becoming paused
since, once again, there is no monitor attached to the callee's channel.
So for the purposes of your setup, the only way you're going to be able to get
what you want working, short of actually changing the source code, is to only
allow the caller to be able to pause the monitor.
Mark Michelson
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Re: [asterisk-users] Override sip.conf settings in extensions.conf? Possible?
Josh Fuller wrote: Hi all... Does anyone know if it is possible to override sip.conf settings in extensions.conf (for example: session-minse=90) without needing to create an overarching peer in sip.conf and selecting it specifically in the dial plan? You can do this to some extent starting with Asterisk 1.6.1. With the AST_CONFIG function, you can change a configuration file from the dialplan. The problem is that you would also have to reload the configuration file so that the change would take effect. After the call was completed, you would then have to reset the value of the option and reload the config file again, since you only want the option set for one call. If this doesn't sound absolutely horrible to you and you want the same functionality in Asterisk 1.4, you may be able to get away with simply copying func_config.c from Asterisk 1.6.1 into Asterisk 1.4's funcs/ directory. I haven't tried this myself, so I don't know what tweaks, if any, would be required to make the code compile. I'm on the 1.4 stable code base and looking to implement session-timers on certain call flows in a modular dial plan. (Sorry if I'm not making the correct logical leap here) Being able to set the session-timers variables via the dialplan will not be sufficient in 1.4 in order to enable session timers on certain calls. You would also have to modify chan_sip.c so that the Asterisk would understand the concept of session timers and how to properly behave. Mark Michelson Thanks, Josh Fuller josh.ful...@telus.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Change Termination of Read Command
Daniel Hazelbaker wrote:
On Apr 27, 2009, at 10:29 AM, Danny Nicholas wrote:
Greetings all,
This is a “just-for-fun” question. I was reading
the support forum and a fellow there wanted Read() to stop on *
instead of #. I thought that changing app_read.c would resolve this
current
if (tmp[x-1] == '#') {
tmp[x-1] = '\0';
break;
new
}if (tmp[x-1] == '*') {
tmp[x-1] = '\0';
break;
}
He applied and recompiled, but no joy. Any ideas why?
Without knowing where in the file this came from I can't say for sure,
but that code looks to me like the code that would run after the digits
are received and is stripping off the # character at the end, if it is
there. Further up (or somewhere else entirely) there is probably a spot
that actually terminates the read command when # is pressed.
Daniel
Daniel is correct in his analysis. If you want app_read to terminate on a '*'
instead of a '#' then you will need to change the ast_readstring call inside of
ast_app_getdata (which is called from read_exec in app_read). This will have
the
side effect of making other situations use a * instead of a # as well (like
entering voicemail mailbox and entering an agent password).
Mark Michelson
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Re: [asterisk-users] Packet2packet bridging while in sip.conf canreinvite=no
jonas kellens wrote: I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on 'SIP/BT201-088f93e0' -- Executing [...@intern:1] Dial(SIP/GXP1200-088f93e0, SIP/BT201|30) in new stack -- Called BT201 -- SIP/BT201-088faa00 is ringing -- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0 *-- Packet2Packet bridging SIP/GXP1200-088f93e0 and SIP/BT201-088faa00* == Spawn extension (intern, 52, 1) exited non-zero on 'SIP/GXP1200-088f93e0' Why is there this native bridging ? Does this mean that Asterisk is no longer in the middle of it ? It is important to note that Packet2Packet bridging is not the same as native bridging. With native bridging, the audio flows outside of Asterisk between the endpoints. With P2P bridging, the audio comes into the RTP layer of Asterisk but does not pass through the Asterisk core. This allows for Asterisk to intercept DTMF or play warning files to the bridged parties. Also : there is no audio at all ! Just when I put down the phone there's the DTMF-signal that the line is cancelled... SIP debug would probably help. Everything worked well before I edited musiconhold.conf and features.conf (to create a park extension). Looking at your musiconhold.conf file, it looks very much like the sample musiconhold.conf file. I doubt that your changes there would have affected anything. If you say that the problems started when you edited features.conf, then I would suggest that you start undoing the changes you made one-by-one to see if you can find what change it was that caused the problem to occur. [sample configs snipped] Do you need extra info ?? What setting can I have set in musiconhold.conf or features.conf to affect the audiostream between my clients ??? There is nothing you can set in musiconhold.conf to control the media stream. With SIP, the signalling still goes through Asterisk even if the media does not. Even if Asterisk is not in the media path, the endpoints can still signal to Asterisk to play MOH to the other side. Asterisk can accomplish this through reinvites. Also, there is nothing you can set in features.conf to control the media stream. Settings pertaining to the media stream are channel-driver-specific and are thus configured in each particular channel driver's configuration file. As you have already discovered, the setting which forces media onto Asterisk during a SIP call is the canreinvite setting. Mark Michelson Before I could call all my clients, I had musiconhold when putting 'on hold' and I was just figuring out how parked calls worked... Thanks for the help ! Jonas Kellens. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] listen to prompt before bridging call.
Deepak wrote: Hi, Can someone please help to resolve the followinng issue: We would like an asterisk user to call a number and when the called party picks up the phone, we play a message (press 1 to accept call, 2 to reject call). Only when the called party presses 1, do we bridge the call and the two parties can communicate. What we would like though is that the person who makes the call be able to listen to the message press 1 to accept call, 2 to reject call) that is played to the called party BUT not be able to communicate with him untill he presses 1. Is this possible in asterisk using php/agi? Any pointers hightly appreciated. I can come up with an easy way to do part of what you want. If you are using the Dial application, you can use the M option to run a macro on the called channel when he answers. Within that macro, you can play prompts to accept or reject the call. The problem is that this does not allow for the calling party to also hear the prompts. Mark Michelson ___ -- 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] pickupexten *8
Gustavo A Gonzalez wrote: Hello all!, I’ve running asterisk 1.4.23.1 and I need to get working pick up from feature.conf. It does no work, only I can connect but cant send audio over the phone. Is there a bug with this feature?. Thanks for any response! Cheers! Yes there was. Please upgrade to 1.4.24, where the problem has been fixed. Mark Michelson ___ -- 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] __ast_read: ast_read() called with no recorded file descriptor
Greg Kennedy wrote: All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.9 spandsp-0.0.5pre4 The receivefax app works perfectly, ie i am able to receive the faxes, and what not, but these messages are filling up my logs. Any ideas what is causing them. I know i saw a message like 2-3 weeks ago about it, but that guy was having e1 problems as well. This is a pure sip environment at the moment. Any pointers would be appreciated. Please see the following bug reports: http://bugs.digium.com/view.php?id=14723 (About the error message) http://bugs.digium.com/view.php?id=14769 (About Fax stuff) The short answer is that it appears there are many places that call ast_read() when they probably shouldn't. The thing is, the error message is what's new, not the other behavior. In other words, there aren't any new problems, just a new error message that points to problems that have been around a long time, most of which probably aren't that big a deal to begin with. Mark Michelson ___ -- 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] AEL2, BASE64_DECODE and hexadecimal
Philipp Kempgen wrote:
Olivier schrieb:
2009/4/7 Philipp Kempgen philipp.kemp...@amooma.de
Olivier schrieb:
I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in
an
AEL2 file like this :
SendText(${BASE64_DECODE(DQ==)});
Value sent (8 bytes long) is very strange :
Content-Type: text/plain;charset=UTF-8
Content-Length: 8
�ez?==
I doubt you will find a way to properly pass CR or LF to an
application in extensions.(conf|ael) but keep us in the loop.
It's strange how such a silly thing is somehow keeping me from centrally
managing phones forwarding : I can display a phone is forwarded but I can't
gracefully return to previous status ...
BTW (developer's question) is there a reason why SendText() resp.
sendtext_exec() refuses to send zero-length data?
I can't point to any specific reason. I assume that whoever wrote the
application probably thought that attempting to send zero-length data was
pointless and that if no data were passed to the application, it likely was due
to an error by the user.
Mark Michelson
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Re: [asterisk-users] Seg Fault after upgrade to Asterisk 1.6.0.8
M Hulber wrote:
Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault.
Reverted to 1.6.0.6 and back to normal.
--
Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24
EST 2009 x86_64 x86_64 x86_64 GNU/Linux
Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at
2ce1ac0537a8 rip 003e980715a8 rsp 7fff5bf00c30 error 4
Apr 3 11:50:00 asterisk kernel: asterisk[3828]: segfault at
0400 rip 003e980758d9 rsp 7fffd3138ef0 error 4
Apr 3 11:50:04 asterisk kernel: asterisk[3879]: segfault at
0c00 rip 003e980758d9 rsp 7fffde4cf280 error 4
Apr 3 11:50:09 asterisk kernel: asterisk[3927]: segfault at
1c00 rip 003e980758d9 rsp 7fff2fd65b10 error 4
Apr 3 11:50:13 asterisk kernel: asterisk[3973]: segfault at
2ce1ac04f948 rip 003e980715a8 rsp 7fff6c283fb0 error 4
Apr 3 11:50:17 asterisk kernel: asterisk[4022]: segfault at
2ce1ac0486e8 rip 003e980715a8 rsp 7fff4e1d0f00 error 4
Apr 3 11:50:21 asterisk kernel: asterisk[4069]: segfault at
2ce1ac067e28 rip 003e980715a8 rsp 7fff2f3ee120 error 4
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117: 5322 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117: 5372 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117: 5419 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117: 5467 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117: 5514 Segmentation fault (core
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS}
${ASTARGS} /dev/${TTY} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Thanks for the information. Could you open a bug report at
http://bugs.digium.com and upload a backtrace from the core dumps? Instructions
for uploading a backtrace can be found in doc/backtrace.txt in the Asterisk
source.
I suspect this is a regression introduced between 1.6.0.6 and 1.6.0.7 since
1.6.0.8 is exactly the same as 1.6.0.7, except for the security fix for
AST-2009-003.
Mark Michelson
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Re: [asterisk-users] Queues in memory after startup
Gabriel Ortiz Lour wrote: Hi all, After * starts the command queue show would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any QueueMemberStatus for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a call? Thanks, Gabriel Ortiz There is a bit of a hack you can use. If you instead use the command queue show some_specific_queue_name then Asterisk will load from realtime. Then type queue show again and you'll see all the queues. I'm not sure why it was written this way. If you use any 1.6 version of Asterisk, you will find that it does not behave this way. queue show will always show all queues. Mark Michelson ___ -- 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-limit=1 breaks attended transfer
carl Lougher wrote: Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff.. Yes, set call-limit to something else :P Seriously though, there's no fix for that since it is behaving exactly as it should. When attempting to transfer the call, Asterisk has no way of knowing that the new SIP INVITE it receives (in order to call the transfer target) is an attempt to transfer the call. It appears that the same SIP peer is attempting to make a second call. Since the call-limit is set to 1, Asterisk rejects the second call attempt. I haven't tried this yet, but it may actually be possible to use DTMF transfers when the call limit is that low since Asterisk is the one that actually initiates the new call to the transfer target instead of the transferer's phone. To use DTMF transfers, you need to set a DTMF sequence in features.conf and use the 't' or 'T' flag (depending on which party should have the ability to transfer the call) in your calls to Dial() or Queue(). Why do you have the call-limit set to 1, anyway? Mark Michelson ___ -- 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] Know who's logged in
Mr. James W. Laferriere wrote: Hello Mark Miquel , On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL Just typing the word agent will result in the message you see. If you press the tab key after typing the word agent you should see that one of your options for completing the command is agent show. This command is definitely in all releases of 1.4. I specifically double-checked and the command works fine for me in 1.4.22. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Diversion header
Olivier wrote: Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used Diversion header is an outdated draft and anyone who follows along with developments in the SIP community will tell you that other methods such as history-info are preferred over use of the diversion header. That being said, in practice, the diversion header is used by several phones. The firmware on my Polycom desk phone (IP 430) supports the sending of a Diversion header when it sends a 3XX response code. As far as Asterisk is concerned, current released versions (All 1.4 and 1.6.0) will read the Diversion header in an incoming response and use that information to fill in the rdnis of the corresponding channel's callerid structure. Once the changes from http://reviewboard.digium.com/r/201 are merged into Asterisk trunk, then Asterisk will also generate a Diversion header if you have configured Asterisk to generate redirecting information. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Diversion header
Olivier wrote: 2009/3/27 Mark Michelson mmichel...@digium.com mailto:mmichel...@digium.com Olivier wrote: Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used Diversion header is an outdated draft and anyone who follows along with developments in the SIP community will tell you that other methods such as history-info are preferred over use of the diversion header. OK, I see that RFC4244 relates to history-info. I'm adding this here for reference. That being said, in practice, the diversion header is used by several phones. The firmware on my Polycom desk phone (IP 430) supports the sending of a Diversion header when it sends a 3XX response code. As far as Asterisk is concerned, current released versions (All 1.4 and 1.6.0) will read the Diversion header in an incoming response and use that information to fill in the rdnis of the corresponding channel's callerid structure. Once the changes from http://reviewboard.digium.com/r/201 are merged into Asterisk trunk, then Asterisk will also generate a Diversion header if you have configured Asterisk to generate redirecting information. Is it planned to support in Asterisk both history-info and diversion headers ? (I can't access reviewboard at the moment soI can't check by myself). If positive, that would be interesting to know how mixed diversion/history-info hardphones are treated. Are you aware of history-info enabled (hard or soft) phone ? I haven't done a lot of research with regards to history-info, so I don't know much about which phones support the feature. I don't know of any immediate plans to place history-info support into Asterisk. Of course, if the core developers were bombarded with requests to add the feature or if a community member were to write support for it into Asterisk then that would increase its consideration for inclusion. Mark Michelson ___ -- 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] Know who's logged in
Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson ___ -- 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] Overriding Queue Wrapup Time
Robert Broyles wrote: Is there a way to override the queue wrapup time on the fly? I would like to allow a longer wrapup time for my agents, but if they are already done with closing up the call ticket, I would like them to be able to dial an extension or something to override the wrapup. Is there a way to do that? There's not a way to do that using the wrapuptime of a queue member, but there are other ways you could potentially take care of this. For instance, you can pause a queue member once he has finished talking and set a timer so that the member will automatically become unpaused after a certain time. If the member is ready to receive calls again before the time has expired, he can dial an extension to unpause himself. Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
Kevin P. Fleming wrote: Mark Michelson wrote: You can work around the bug, although it's not exactly optimal. What you can do is to modify your dialplan as follows: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) Couldn't you just set _DYNAMIC_FEATURES here and have it get automatically inherited to the outbound channel? Yes, that would be another suitable workaround. I sometimes forget that you can let variables inherit like that. Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: The patch doesn't work for me. Here's what I did: Changed to my asterisk-1.4.23.1 directory Executed the wget / patch command from the link you provided make saw that res_features.so was recompiled Moved /usr/lib/asterisk/modules/res_features.so to res_features.so.old make install Confirmed that res_features.so was recreated in /usr/lib/asterisk/modules asterisk -r -- I never shut asterisk down module unload res_features.so module load res_features.so After this there was no change, it worked using the macro but using the Set(DYN... on the caller only. Thanks, All right. Let's continue this discussion on the bug report I opened. To start with, could you upload console output from an attempt at using the dynamic feature with my patch attached? For the console output, it would help if the verbose and debug levels were both set to at least 4. That way I can hopefully see what the problem is. Thanks. Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: I'm sorry, but it looks like it's working correctly now. I will update the bug if I am able to verify any problems. Thanks, Heh, no reason to be sorry for it working :) When you say it works now, was this with or without the patch applied? Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: It was with the patch applied, but after I restarted asterisk. Thanks, Fix committed to Asterisk 1.4 in revision 181990. Mark Michelson ___ -- 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] Are .call files working with extensions.ael ? bristuff problem
Peer Oliver Schmidt wrote:
Olivier wrote:
Do you by chance use bristuff?
Yes, I do.
bristuff patches pbx/pbx_spool.c
I have no knowledge of C, but there seems to be a problem around line
266.
The original line (pre-bristuff) looks like this:
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten))) {
The patched line looks like this:
if (ast_strlen_zero(o-tech) || ast_strlen_zero(o-dest) ||
(ast_strlen_zero(o-app) ast_strlen_zero(o-exten)) ||
(ast_strlen_zero(o-message) ast_strlen_zero(o-pdu))) {
Try reverting that line, and see if that helps with your problem. And
maybe someone with a better understanding of C can take a look at the
above problem.
Apparently bristuff has added new required parameters to call files. Basically,
it has the same requirements as vanilla Asterisk (you must specify a full
channel name and either an app or an extension) and it also requires that one
of
the message or pdu fields of the outgoing call are filled in. You may want
to check bristuff documentation to figure out what these mean since they are
not
part of a regular Asterisk installation.
Mark Michelson
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Re: [asterisk-users] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: I'm trying to actually use the example application map in features.conf: testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play ;tt-monkeys to the opposite channel I see the feature get registered at the CLI: == Registered Feature 'monkey' == Mapping Feature 'monkey' to app 'Playback(tt-monkeys)' with code '#9' But I'm unable to actually use it. This *doesn't* work: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) exten = 301,n,Dial(SIP/DavidR1) Anyone done this before and/or able to give me any suggestions? Thanks, I strongly suspect that you have fallen prey to the featuredigittimeout. Check your features.conf file for the featuredigittimeout option. By default, this is set to 500 ms. You probably want to increase this to something like 2000 ms. This option specifies the amount of time Asterisk should wait between DTMF presses when you are dialing a feature code. So in your case, I'm guessing that you pressed # but could not press 9 in time for Asterisk to recognize this input as part of the same feature. Mark Michelson ___ -- 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] Timeout for Queue
Darrin Henshaw wrote: Hello, We had an incident recently where a call was in queue for an extended period of time. We use queuemetrics for reporting, and it reports that the call was waiting for 20 minutes. The different thing about it is that the disconnect reason is stated as Timeout. Is there a set maximum time a call will wait in the queue before being automatically disconnected? I tried looking through the code directly, but I humbly admit my programming skills are lax. I’m running Asterisk 1.2.31 on CentOS 4.7. Thanks. A brief disclaimer: I admittedly don't have much experience with Asterisk 1.2, but I don't *think* this behavior is different between 1.2 and any later versions. What I'm stating below holds true for Asterisk 1.4.X and Asterisk 1.6.X The Queue application has no default timeout. A caller can stay in the queue forever if no timeout was given when calling the Queue application. Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: I don't really think that's a problem, because I'm able to use the other built in options: *1 to record; ## transfer (I changed this from a single pound) and there have been a couple times that I wouldn't hit them quickly enough. Thanks, Ah, sorry about that. The featuredigittimeout burns so many people that it's pretty much a knee-jerk reaction on my part now to suggest that as a potential fix. To test out, I set up the same feature and gave it a try with a current subversion checkout of Asterisk 1.4. I placed a call from SIP/2001 to SIP/2000 and here's what I found. When SIP/2001 pressed #9, tt-monkeys played on SIP/2000's channel When SIP/2000 pressed #9, nothing happened. I tried modifying the features.conf line to have peer/callee instead of just peer and that caused neither side to successfully use the dynamic feature. There appears to be a bug which does not allow for the callee to use dynamic features. The problem appears to be that when DTMF is pressed, we try to interpret the presses to determine if there is a corresponding feature. The DYNAMIC_FEATURES variable has been set on the caller's channel, but has not been set on the callee's channel. As a result, we don't properly read the value of the DYNAMIC_FEATURES variable if the callee is the one to press DTMF. You can work around the bug, although it's not exactly optimal. What you can do is to modify your dialplan as follows: exten = 301,n,Set(DYNAMIC_FEATURES=monkey) exten = 301,n,Dial(SIP/DavidR1,,M(dynamic_features)) [macro-dynamic_features] exten = s,1,Set(DYNAMIC_FEATURES=monkey) By doing this, the dynamic_features macro will be called on SIP/DavidR1 when he answers. This will allow for the DYNAMIC_FEATURES variable to be set on both channels so both sides can use the feature you have set. This is a bug, and so there needs to be action to fix it correctly. What I've suggested is just a workaround, but it should get you through your problem for now. Mark Michelson ___ -- 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] Trying to get sample applicationmap to work (*1.4)
David Ruggles wrote: Wow! Thanks! That's a very clear answer and completely understandable. Is this something I should open a bug report on? Thanks, Nope, I've already got that taken care of. http://bugs.digium.com/view.php?id=14657 There's a patch there that I have tested and it works for me (TM). See if it works out for you, too. Mark Michelson ___ -- 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] 1.6.x differences
Joseph L. Casale wrote: What are the differences, or where do i find docs on the difference between the 1.6.0.x and 1.6.1.x release? Thanks! jlc A good place to find that out is to look at the CHANGES file in the Asterisk source. This file tells the of new features/behavior added since the previous version. In addition, you can check UPGRADE-1.6.txt to see about changes you may have to make when upgrading. Mark Michelson ___ -- 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] chan_zap.so missing
Vieri wrote: --- On Tue, 3/10/09, markus antro...@googlemail.com wrote: Now I am missing /usr/lib/asterisk/modules/chan_zap.so. I searched through the mailing list and forums. They say, that chan_zap.so is build in channels/ in my working directory. But it's not too strange, that chan_zap.so was not built, since there's not even the file chan_zap.c . Isn't it supposed to be there? Sorry to barge in like this but I would like to know if chan_zap.c is supposed to be present in 1.4.23.1. Vieri The actual file chan_zap.c is not in 1.4.23. It is called chan_dahdi.c instead, BUT using Zaptel with 1.4.23 is fully supported. In addition, the use of channel names like Zap/1 is supported in 1.4.23 as well. In fact, to make the policy a bit more clear, all releases of 1.4 will work with Zaptel and should not require any special configuration changes when upgrading to a version with DAHDI support. All variants of Asterisk 1.6 only support the use of DAHDI. Mark Michelson ___ -- 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] Fwd: add a new queue strategy: SBR
nik600 wrote:
On Mon, Mar 9, 2009 at 3:16 PM, James Sneeringer jsnee...@gmail.com wrote:
If you are using dynamic queues with Local channels (as described in
doc/queues-with-callback-members.txt in the Asterisk source), you can
also optionally implement this functionality directly in the dialplan.
This has the added benefit of allowing you to choose on a per-agent
basis who is eligible for autopause.
-James
thanks for your reply, infact i've implemented the agents in the
dialplan as explained in queues-with-callback-members.txt but this
approach doesn't manage the status of the agent!
I can add / remove / pause / unpause the member interface but what
about the in use status?
The extension in the context will be every time Not in use or shall
i implement hints?
Here there is a piece of my extensions.conf:
[default]
; login procedure for queue 001
exten = _001,1,Answer
exten = _001,n,AddQueueMember(001,Local/${EXTEN:3...@agents)
exten = _001,n,Set(DB(agents/${EXTEN:3})=SIP/${CALLERID(num)})
[agents]
exten = _,hint,${DB(agents/${EXTEN})}
exten = _,1,Dial(${DB(agents/${EXTEN})})
and there isn't an agent but only an extension on a queue.
What do you think about that?
maybe i should open a new post but i think that this kind of approach
isn't much better than the callback functionality, what do you think
about that?
The reason that the member always appears to be not in use is that local
channels are optimized away once they are bridged to their real destination.
The
result of this is that since the channel does not exist anymore, the device
state engine interprets the interface to be not in use anymore. One way to
handle this issue is to change your AddQueueMember call to use
Local/${EXTEN:3...@agents/n (notice the /n at the end). The /n tells the local
channel driver to not attempt to optimize the local channel away.
If you are using Asterisk version 1.6.0 or above, an even better method would
be
to specify a second interface to poll for device state when adding the queue
member. Assuming that the member at Local/${EXTEN:3...@agents will always call
SIP/${EXTEN:3}, then what you are really interested in when receiving device
state notifications is the SIP channel, not the local channel. You can specify
this second state interface in AddQueueMember like so:
AddQueueMember(001,Local/${EXTEN:3...@agentsSIP/${EXTEN:3})
Doing this will tell app_queue to use the SIP channel's device state to
determine if the member is available, but when it comes time to call the agent,
it will actually place the call to the local channel provided.
Mark Michelson
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Re: [asterisk-users] MoH - always starting from the beginning
Mike wrote: Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so that it just plays like a CD and the person hears wherever the stream is at at a given moment? Regards, Mike What version of Asterisk are you using? There was a recent bug introduced in 1.4.23. The fix for the issue is here: http://svn.digium.com/svn-view/asterisk?view=revrev=174218 Mark Michelson ___ -- 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] MoH - always starting from the beginning
Mark Michelson wrote: Mike wrote: Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so that it just plays like a CD and the person hears wherever the stream is at at a given moment? Regards, Mike What version of Asterisk are you using? There was a recent bug introduced in 1.4.23. The fix for the issue is here: http://svn.digium.com/svn-view/asterisk?view=revrev=174218 Mark Michelson After my message got posted, I realized that my message was very nonspecific, and I should have linked to the originally filed bug report. Here's a link to it: http://bugs.digium.com/view.php?id=14407 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cdr problem
Anthony Francis wrote: Tilghman Lesher wrote: On Friday 06 March 2009 11:24:46 pm Hooman Peiro wrote: hi, I'm working with asterisk on a project and I found a problem with cdr_odbc. As we know, after answering each call a cdr event is raised which is saved in cdr_csv and cdr_odbc. but here my point is on cdr_odbc. some information, including start_time and end_time is given by cdr event but the problem is that these two information(start_time and end_time) is not getting save in cdr_odbc. I checked the source code and I found that by default it's not doing so. I need to query these two information, start time and end time, from cdr_odbc and I need your help. thanks You are partially incorrect. The start time is indeed stored in the CDR, although the column name is 'calldate'. As for the end time, it can be derived by adding 'duration' (which is in whole seconds) to the 'calldate' column. Another solution that allows for retrieving both columns with their native names (or completely different names, whatever you map it to) is to use cdr_adaptive_odbc in 1.6.0 and higher. I have often thought, wouldn't it be better if the cdr config files allowed you to specify column names i.e. calldate = callstart_datetime Or whatever, the basic format being asteriskfieldname = db columnname. Just an idea.. Anthony Francis This suggestion of yours is exactly what cdr_adaptive_odbc that Tilghman suggested does. If you're using Asterisk 1.6.0 or higher, take a look at configs/cdr_adaptive_odbc.conf.sample for some examples. Mark Michelson ___ -- 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] Fwd: add a new queue strategy: SBR
David fire wrote: you are wrong. when you set up an agent in a queue you can put a priority. David The term used in Asterisk for a queue member's priority is the word penalty. When you set up a member in queues.conf, the penalty is the third option for a member. Here's an example: member = SIP/2000,Mark Michelson,3 In the above example, Mark Michelson is the name of a queue member who can be reached by calling the interface SIP/2000. His penalty is 3. The rule for penalties is that members with lower penalties are called before members with higher penalties. If all the members of the lowest penalty are unavailable (i.e. not logged in or currently on a call) then the Queue application will attempt to call a member with a higher penalty. Caution: One shortcoming of queue member penalties is that they are not taken into account if a queue member of a low penalty does not answer a call. Say for instance that the queue application determines that there are two members available to answer an incoming call. One member has penalty 1 and the other has penalty 2. If the member with penalty 1 does not answer the call, the queue application still considers that member to be available the next time that it tries to reach a member. The member with penalty 2 will only be tried if the queue application can determine *before the call is placed* that the member with penalty 1 is unavailable. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP *8 Pickup Problem
Klaus Darilion wrote: Hi! I have the following weird problem: phones A,B and C are in the same callgroup/pickupgroup. A call B, B is ringing, C calls *8. Now, B is CANCELed, C gets 200 OK, but A is still in Ringing. Is there anything else I have to configure? thanks Klaus What version of Asterisk are you using? If you're using 1.4.23, there was a confirmed problem which has been fixed now in the 1.4 svn branch. For the issue, please see http://bugs.digium.com/view.php?id=14206 Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP *8 Pickup Problem
Remco Barendse wrote: On Fri, 6 Mar 2009, Klaus Darilion wrote: Updating to 1.4 branch solved the issue. Thanks. Pity that they still didn't release a new version that works properly. We can't afford to release a new version every time we fix a bug. That's just not practical. 1.6.0.6 is broken too, SIP doesn't work on 2 difference boxes i tried it on. What's broken exactly? Saying SIP doesn't work is not a helpful description of what is going wrong. Are there any open bug reports that describe the problem you are having? Mark Michelson ___ -- 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] GoSub Queue
Shaun R. wrote:
I have a caller screen queue setup. Basically a caller calls in, goes
through a IVR, and before that caller is put into the queue, they get a sub
ran on them first asking for them to say there name. That gets saved and
they are entered into the queue using Queue(mainqueue300).
In the queues.conf i have a list of members these are
local/extens...@external-default, there are two weights/prioritys 10 and 20.
The external-default context has a dial that uses a GoSub to play the
recording of the caller to the member, it them gives the member a list of
options like Connect, Voicemail, Hangup.
The problem i'm having right now is that if a member pics up, all the phones
of the other members continue to ring (ringall in queues.conf) while the
member who answered the phone listens to the announce. Once he chooses a
option they stop. If he doesnt choose a option and lets those phones dial
time out then they stop ringing but whats weird is that queue doesnt go and
ring the next priority. I dont want it to ring the next priority but i find
it weird that the queue knows that sombody has the call, but doesnt stop
rining those other extensions.
Now here's another way i found to do this, rather than using dial with a
gosub i found that i can put the gosub as part of the queue() command. [
example: Queue(mainqueue300,,,screencallee) ]. This will run that gosub
when the member pics up and it DOES stop ringing all the other phones when
they pickup which is great! problem is that now my options like Voicemail
or if the member hangs up with out choosing a option the call is dumped from
teh queue and the GOSUB_RESULT doesnt look to be checked or listened too.
Here's a snip of my screen subs
[screencaller]
exten = s,1,Set(__SCREEN_FILE=/tmp/screens-${UNIQUEID})
exten = s,n,Playback(screen-say-name)
exten = s,n,Wait(1)
exten = s,n,Record(${SCREEN_FILE}.gsm,3,6)
exten = s,n,Playback(screen-please-hold)
exten = s,n,Return
[screencallee]
exten = s,1,answer
exten = s,2,Set(GOSUB_RESULT=CONTINUE); set default to continue so that if
something funky happens the call is returned to the queue
exten =
s,3,background(screen-call-for${SCREEN_CALL_TO}from${SCREEN_FILE}screen-press-1-connectscreen-press-2-holdscreen-press-3-voicemail/screen-press-4-hangup)
exten = s,4,WaitExten(20)
;; Connect the caller
exten = 1,1,Set(GOSUB_RESULT=)
exten = 1,2,Return
;; Put Caller on Hold
exten = 2,1,Set(HOLD_LOOPCOUNT=0)
exten = 2,2,read(HOLD_OPT,screen-on-hold,1,,1,5)
exten = 2,3,GotoIf($[${HOLD_OPT} != ]?s,1)
exten = 2,4,GotoIf($[${HOLD_LOOPCOUNT} 10]?h,1)
exten = 2,5,Set(HOLD_LOOPCOUNT=$[${HOLD_LOOPCOUNT}] + 1])
exten = 2,6,Goto(2,2)
;; Send the caller to voicemail
exten = 3,1,Set(GOSUB_RESULT=GOTO:voicemail^${SCREEN_VM_EXT}^1)
exten = 3,2,Return
;; Hangup on the caller
exten = 4,1,Set(GOSUB_RESULT=GOTO:hangup^s^1)
exten = 4,2,Return
;; Return Caller to queue
exten = 5,1,Set(GOSUB_RESULT=CONTINUE);
exten = 5,2,Return
;; Reidentify the caller
exten = 6,1,Goto(s,1)
;; Invalid Option
exten = i,1,Playback(invalidoption)
exten = i,2,Goto(s,1)
;; Timeout Reached :: Hangup Called Party, Return Callee to Queue
exten = t,1,Set(GOSUB_RESULT=CONTINUE)
exten = t,2,Noop(GOSUB: screencallee timed out)
exten = t,3,Return
;; Hangup :: Hangup Called Party, Return Callee to Queue
exten = h,1,Set(GOSUB_RESULT=CONTINUE)
exten = h,2,Noop(GOSUB: screencallee hangup)
exten = h,3,Return
Any help?
~Shaun
The problem is that several parallel instances of the Dial application are
happening at the same time on the local channels. None of these parallel
instances know of any of the others, so they continue to ring even though one
of
the Dials has been answered.
Right now there's nothing that can be easily done about this from your end,
unless you have the ability to not use local channels for your queue members.
In writing this response, I wonder how easy it would be to make that answer in
your gosub get pushed up to the queue application so that it can hang up the
other local channels it has dialed. I think I'll set this scenario up and see
what I can find out.
Mark Michelson
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Re: [asterisk-users] question about ringinuse
Sebastian wrote: Just a silly question that I’m not sure. Ringinuse is working with IAX in 1.6??? like in sip?? I assume you're referring to the queues.conf option, correct? An easy way to check is to issue a queue show command when an IAX2 queue member receives a call. If his status is in use then ringinuse should work correctly. If the device state is reported as anything other than in use then that would indicate that IAX2's device state reporting may be inaccurate and also ringinuse will not work. Mark Michelson ___ -- 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] What's the use of sip.conf's notifyringing ?
Olivier wrote: Hello With 1.4.23.1, I can't really see any difference between setting this value to yes or no. Can you explain ? Regards It seems that you're only going to see a difference if you are using a phone which subscribes to hints and uses the application/dialog-info+xml event package (comments in the code suggest that SNOM phones use this method). If notifyringing is set to yes (or if the option is not specified at all since yes is the default value) then when Asterisk sends a NOTIFY to a phone, it will set the information enclosed in the state XML tag to ringing instead of confirmed if a phone is ringing. Also, Asterisk will place a direction attribute inside the dialog XML tag in this situation too. The short version of this is that the notifyringing option will specify a ringing state in NOTIFY messages but only for certain types of phones. Mark Michelson ___ -- 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 generate core dump?
Ken D'Ambrosio wrote: Asterisk segfaulted on me the other day; how do I tell it to generate a core file so -- if it happens again -- I can attempt to debug? I looked in the obvious places in make menuconfig and didn't see anything appropriate. Thanks, -Ken Run Asterisk with the -g option and it will dump a core file if it should crash. Mark Michelson ___ -- 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] What is the purpose of membermacro in queues.conf
Rajkumar S wrote: Hi, There are 3 new settings (setinterfacevar, setqueueentryvar, setqueuevar) and membermacro settings in 1.6 queues.conf. What is the potential use of these settings? The variables set are useful, but there is no indication of the purpose they could be used? Any one with some light on potential use case of these new features? raj I'd be glad to explain them. First of all, setinterfacevar was actually around in 1.4, but its use has been expanded in 1.6.0. In 1.4, this would cause the MEMBERINTERFACE channel variable to be set. In 1.6.0, this setting also sets the MEMBERNAME, MEMBERCALLS, MEMBERLASTCALL, MEMBERPENALTY, MEMBERDYNAMIC, and MEMBERREALTIME variables. The purpose of exposing these values is to allow for an administrator to use these for any purpose he may desire. Second, there's setqueuevar. Its purpose is similar to setinterfacevar, in that it exposes values to the dialplan so that an administrator can use them how he wishes. The variables set are QUEUENAME, QUEUEMAX, QUEUESTRATEGY, QUEUECALLS, QUEUEHOLDTIME, QUEUECOMPLETED, QUEUEABANDONED, QUEUESRVLEVEL, and QUEUESRVLEVELPERF. Finally, you asked about membermacro. This allows for a macro to execute on a queue member's channel when he answers the call. This is very similar to the 'M' option for the dial application. Some people use this sort of feature as a post-answer hook into the dialplan so that they can perhaps log statistical information, or present the queue member with information about the incoming call. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding custom announcements in queues.conf
Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson ___ -- 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 regarding custom announcements in queues.conf
Mark Michelson wrote: Christopher Aloi wrote: Hey List, Anyone know the correct way to override an announcement on a queue by queue basis? My goal is to have one of my queues say press one to blah.. and no position announcements I have the jump from queue context working (the press 1) I just need the correct message played to the user instructing to press 1. I have periodic-announce=filename in my queues.conf file under the correct queue, but queue-periodic-announce is played to the caller, not my custom file. Here's the queue listed in queues.conf: [EXAMPLE-QUEUE] maxlen=20 reportholdtime=no periodic-annouce = SD-PLS-HOLD periodic-announce-frequency=10 announce-holdtime=no strategy=ringall joinempty=yes retry=5 timeout=30 music=CUSTOM autofill=yes context=queue-jump member = SIP/7909416...@192.168.13.32 mailto:7909416...@192.168.13.32 When the call comes into this queue after 10 seconds the following occurs: -- Stopped music on hold on SIP/100-FOO-b781a4c0 -- Playing periodic announcement -- SIP/100-FOO-b781a4c0 Playing 'queue-periodic-announce' (language 'en') What can I do to make this play the SD-PLS-HOLD wav I defined above? Thanks! A quick look at the code and your config leads me to believe you're doing everything correctly. What version of Asterisk are you using? Are you using realtime queues/queue members? Mark Michelson Hmm, my realtime question is a bit silly since you provided config for a static queue with a static member in it. My question about the version is still relevant, though. Mark Michelson ___ -- 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 regarding custom announcements in queues.conf
Christopher Aloi wrote: Here's the version - Asterisk SVN-branch-1.4-r143404 Just static queues. Is it true that Asterisk looks in the default /var/lib/asterisk/sounds/ dir for these queue announce files? So my custom file should live in that dir right? Thanks for the help :) Yes, if an absolute path is not provided for the sounds, then it is assumed that the default sounds directory is where the sound may be found. I just tried a small test on that revision of 1.4, and it worked for me. In my case, I was simply playing the beep sound file which already exists in the sounds directory. Mark Michelson ___ -- 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 regarding custom announcements in queues.conf
Christopher Aloi wrote: Yah - Found my problem, I can't spell - periodic-*annouce* = SD-PLS-HOLD periodic-announce-frequency=10 : ) Oh, Ha! That'll do it every time. Mark Michelson ___ -- 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] command show channels concise
Jerry Geis wrote: I am getting a priveldged command error on the manager API. 16-Feb-09 11:51 am asterisk_command() Action: Login 16-Feb-09 11:51 am asterisk_command() Username: XXX 16-Feb-09 11:51 am asterisk_command() Secret: 16-Feb-09 11:51 am asterisk_command() Events: off 16-Feb-09 11:51 am DEBUG: Response: Success[CR ][LF ]Message: Authentication accepted[CR ][LF ][CR ][LF ] 16-Feb-09 11:51 am asterisk_command() Action: Command 16-Feb-09 11:51 am asterisk_command() Command: show channels concise 16-Feb-09 11:51 am DEBUG: Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ] manager.conf has: [XXX] secret= permit=127.0.0.1/255.255.255.0 read = system,call,command,all,agent,user write = system,call,command,all,agent,user I thought that was all I needed to run that command? I am using 1.4.23. Jerry That's not an error message. That is the response given to a Command action assuming that a command was provided and the command is not blacklisted. Mark Michelson ___ -- 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] reinvite
Jeff LaCoursiere wrote: On Mon, 9 Feb 2009, Jeff LaCoursiere wrote: I've never used reinvite in systems I have installed to date, and I have finally run across a situation where it would be preferred. A remote office has a flaky Internet connection. With G729 encoding the calls to the central office over the 'net are tolerable. One Linksys 2102 drives two phones at this location, and when the first one calls the second one it travels to the central office and back, which is no longer tolerable. For each sip peer I have canreinvite=yes, but I am a bit confused as to the correct options on the 2102 to use this feature. Is anyone doing this with 2102s that can give me some pointers? I have been playing around with this in my lab and cannot seem to make it work as expected. I have a remote asterisk server on a public IP - 1.4.22-3 on Centos 5. I have two Polycom IP501s on a local LAN behind a NAT gateway. Both Polycom's register with the remote server and can call each other without issues. Both SIP contexts have nat=yes, canreinvite=yes. The caller is 223, the callee is 222. eth0 is the outside (public) interface, XXX is my dynamic IP. I trapped a conversation on the asterisk server with: tcpdump -nli eth0 -s 0 -w /tmp/reinvite.debug host XXX and not port 22 While this was running I made a call between the two extensions for a few seconds then hungup. I opened this capture in etherreal and can see the following: 223-AST INVITE 2...@ast AST-223 407 Proxy auth required 223-AST ACK 223-AST INVITE 2...@ast, with proxy-auth info AST-223 100 Trying AST-223 200 OK 223-AST ACK Then I see the RTP traffic begin back and forth. I am confused on two fronts - first where is the INVITE from AST to 222? Not sure how I missed capturing that side of the conversation. And of course where is the AST reinvite? It isn't occurring since I can clearly see the RTP traffic flowing via the asterisk server. Any ideas? Cheers, j Asterisk may not be sending reinvites to the phones due to options you have passed to the Dial application. If Asterisk needs to intercept DTMF for a feature, then Asterisk will not send reinvites to the endpoints to redirect the media. For instance, if you have the 't' or 'T' options enabled in your Dial application, then Asterisk will not send reinvites to the endpoints even if you have configured chan_sip to allow reinvites to be sent. Other factors which can contribute are use of applications like Monitor and MixMonitor which require the media to go through Asterisk. Mark Michelson ___ -- 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] 1.6.1-rc1 errors
Carlos Chavez wrote: I am getting the following warnings on the CLI when loading Asterisk 1.6.1-rc1: [Feb 12 12:32:34] NOTICE[22261]: timing.c:59 ast_install_timing_functions: Multiple timing modules are loaded. You should only load one. [Feb 12 12:32:34] ERROR[22261]: codec_dahdi.c:398 find_transcoders: Failed to open /dev/dahdi/transcode: No such file or directory [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10939 build_user: Unable to support trunking on user 'telecomab' without DAHDI timing [Feb 12 12:32:33] WARNING[22261]: chan_iax2.c:10679 build_peer: Unable to support trunking on peer 'telecomab' without a timing interface I am using DAHDI 2.1.0.4, Asterisk 1.6.1-rc1 on a CentOS 5.2 machine with a TDM04 card. These are the modules: Module Size Used by dahdi_echocan_mg2 9608 0 wctdm 39884 4 dahdi 190728 2 dahdi_echocan_mg2,wctdm Where do I have to specify the timing module? Timing may be provided from one of two sources in Asterisk 1.6.1: res_timing_dahdi.so (get timing from DAHDI), and res_timing_pthread.so (use pthread library for timing). There are a couple of ways to fix your problem, assuming that the timing module you want to use is res_timing_dahdi.so. 1) Remove res_timing_pthread.so from /usr/lib/asterisk/modules and restart Asterisk 2) In modules.conf, add noload = res_timing_pthread.so 3) While not a requirement, you can also make menuselect and disable res_timing_pthread.so from being built at all. The module can be found under the Resource Modules menu. It looks as though the timing modules for 1.6.1 are not well-documented, and Menuselect should be altered to not allow for both modules to be built. We'll get to work getting this documented better. Mark Michelson ___ -- 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 and SIP transfers (SIP REFER)
Gunnar Schaller wrote: Hello list, I need to record all calls. So I'm using application Monitor. Works good until someone transfers a callee to another internal extension. Example: A calls B A set B on hold A calls C A transfers B to C with SIP transfer (SIP REFER - with phone funktions and not Asterisk attended transfer). I found http://bugs.digium.com/view.php?id=0013538 . corruptor asked about this problem, but it seems there is no solution. Now I want to know how anyone deals with this problem. How to record those transfered calls? Any solution with manager commands or some source-code hacking (enabling Monitor for all calls so no Monitor is needed in dialplan). I'm working with Snom phones here - so there is the possibility to work with action url's. Thank you, Gunnar The problem in this particular case is that the actual monitor object is on A's channel. When A is no longer involved in the call, the monitor is gone, and so the call cannot be recorded further. One possible solution is to run the Monitor application on B's channel instead. This can be done by using the M option in the Dial application. The M option allows you to run a macro on the *called* channel's party when he answers. If you start the Monitor application from this macro, you should find that things will work as you expect. Note that the issue you linked was about MixMonitor, not Monitor. They are completely different beasts when it comes to how they operate. In fact, MixMonitor recordings can be set to survive a transfer if you are using Asterisk 1.4.23 and make use of the AUDIOHOOK_INHERIT function. For more information on its use, you can issue the command core show function AUDIOHOOK_INHERIT from the Asterisk CLI. Mark Michelson ___ -- 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] upgrade from 1.4.22-rc5 to 1.4.23.1: crash when transferring a call
Giorgio Incantalupo wrote: Hi, just upgraded my Asterisk from 1.4.22-rc5 to 1.4.23.1 keeping the same zaptel/libpri/mISDN/add-ons. It crashes when transferring a call. Anybody tried it with success? Thank you Giorgio If you're having crashes occur when transferring a call, you should report it as a bug on bugs.digium.com. Be sure to attach a backtrace from the crash as described in doc/backtrace.txt in the Asterisk source. Thanks, Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie query: how to write priority n+101
Geoff Lane wrote:
Hi All,
Asterisk 1.4.12 on CentOS 5
Sorry for a question that I'm guessing is obvious to most of you.
I'm trying to revamp my dialplan. When I first created it, I had
something like:
exten = s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten = s,2,Dial(${rgMain},${RINGTIME},t)
exten = s,3,VoiceMail(m...@default)
exten = s,103,VoiceMail(m...@default)
Now I want to play around to add things like the privacy manager and
blacklist handling, which all goes before priority 2 in the above. The
Dial() application jumps to the priority 101 more than its own
priority (i.e. n+101) if it times out. But how can I specify this if
I'm numbering priorities as 1,n,n,n,n?
(BTW, the reason for priority 3 in the above extension is that in an
earlier version of Asterisk, Dial() sometimes jumped to the next
priority rather than one 101 more).
TIA,
Actually, jumping to priority n + 101 is a thing of the past, and this will
only
occur now if you pass the 'j' option to Dial. Dial will just go to the next
priority on a timeout now, and the DIALSTATUS channel variable will be set to
NOANSWER I suspect that if you enable verbose console logging, you'll
actually
see that priority 3 is what is being executed and not priority 103.
Check out the UPGRADE.txt file in Asterisk 1.4. In the Applications section,
you'll see:
* 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'.
Mark Michelson
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Re: [asterisk-users] Newbie query: how to write priority n+101
Geoff Lane wrote: On Thursday, February 5, 2009, Tilghman Lesher wrote: The correct string is FAILED, not FAILURE. Thanks. For info, *TFOT says: PrivacyManager() sets a channel variable named PRIVACYMGRSTATUS to either SUCCESS or FAILURE. If Caller ID is received on the channel, PrivacyManager() does nothing. I've tried it and you're correct. So it looks like the docs need a bug report - any idea how I go about that? Thanks again, If you're using the 2nd edition of the book, check the preface, page xix for contact information. For those monitoring the mailing list who do not have a copy of the book, the following web page is listed as containing errata, examples, and any additional information: http://www.oreilly.com/catalog/9780596510480 Mark Michelson ___ -- 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] Broken Pipe error while using UpdateConfig command
Remco Barendse wrote: 1.4.23.1 is quite badly broken and there are no significant new features There are no new features at all, actually. What problems are you having with 1.4.23.1? It doesn't accomplish much to say that it is quite badly broken without at least telling what is wrong. We can't fix what's wrong if we don't know what's wrong to begin with. :) Mark Michelson ___ -- 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 set udptl.conf ?
Olivier wrote: Hi, voip-info.org http://voip-info.org is almost silent regarding udptl.conf except with Depending on your fax device (such as the Linksys 3102) you may have to edit the udptl.conf file. The error correction type that is sent is usually the culprit of many problems with ATAs and T.38 providers. Can anyone elaborate on this ? Regards If you look at configs/udptl.conf.sample, all the options are outlined there, although the explanations are very brief. The most likely setting that is referred to on voipinfo is the T38FaxUdpEC setting. Possible values are t38UDPFEC and t38UDPRedundancy, with the former being the default. Looking at the code, it appears that the options are case-sensitive for udptl.conf, which is quite a bit different from the rest of Asterisk, so be sure to get the case correct. Mark Michelson ___ -- 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] Warnings during a compile
Robert Boardman wrote: Here is just one example of a warning when compiling asterisk on Ubuntu 8.10 manager.c:1760: warning: ignoring return value of âreadâ, declared with attribute warn_unused_result is this anything to worry about? can i safely ignore it? Thanks Robb I may be wrong about this part, but that class of warning is something that started appearing with a recent version of gcc (4.3 I think). Kevin Fleming took the time to clear up these warnings shortly after the release of this version of gcc, so if you are using a current checkout of Asterisk, you shouldn't see those warnings. In fact, looking at manager.c in my 1.4 and 1.6.0 checkouts, all calls to read(2) have their return value checked. To answer your question more directly, it's something that has a low potential to create problems, but given how long Asterisk had gone without checking those return values and how recently that was fixed, it's probably something you can ignore. Of course updating to a more recent checkout of Asterisk will clear such warnings up. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Warning in CLI
Mike wrote: Hi, Anyone can tell me what this means? [Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter: Inringing for peer 'test-peer' 0? Regards, Mike A sip_peer object in Asterisk has an inUse and inRinging number associated with it to keep track of the number of lines in use and the number of lines that are ringing for a peer. These numbers should not be less than 0, and if it is, then it indicates a problem has occurred at some point. There was a problem with some early 1.6.0 releases with these counters falling below 0, but as far as I know, that should be corrected in later versions. What version of Asterisk are you seeing this with? Mark Michelson ___ -- 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] ChanSpy or other variant
Nicholas Blasgen wrote: I'm trying to figure out how to listen in to a channel that I specify. I have the impression I've seen this done via Flash web controls, but I'm trying to write something myself and I can't figure out what command would be used. ChanSpy looks great, but I don't see how to specify the channel. I have a channel identifier like SIP/provider-08748db0 which is what I would send to applications like Hangup(chan) or Redirect(chan) but it doesn't look like ChanSpy was written to accept that format. I haven't tried passing SIP/provider-08748db0 to ChanSpy, but from the documentation it seems that it shouldn't work. So the question is, how can I listen into a channel if I know either the channel or the unqiue id? And in the meantime I will play around with ChanSpy more. Chanspy should do exactly what you want. If you ran exten = blah,n,ChanSpy(SIP/provider) Then you would be able to listen to all active calls involving any channel whose name begins with 'SIP/provider'. If it turns out that there is a channel called 'SIP/provider-12345abc', then that channel may be spied on with the above ChanSpy call in the dialplan. The thing to remember is that the chanprefix argument as it is described in ChanSpy's documentation is literally any text that may appear at the start of a channel name. Chanspy(SIP) would allow you to spy on any SIP channel, whereas ChanSpy(S) would allow spying on both SIP and Skinny channels. There is no minimum or maximum limit to what this string may be. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
Matt Florell wrote: Yep, my bad I found them once I searched with the dash '-' after the 1.4.23. They were lost in the flood of users list mail in my inbox. I wonder if these could also be posted on the asterisk-announce list more consistently? I see a few releases on the announce list, but last 1.4 one was December 2nd and nothing after that on that list except for a few vulnerability postings. The policy that we have been following is that only final releases will be announced to the asterisk-announce list. Betas and release candidates are not. The rationale is that asterisk-announce is supposed to be a low-volume list and that most subscribers to it would not appreciate all the noise of announcing release candidates or betas there. I should think that the policy could be amended; however, I'm not really in a position to make that call, nor do I know if you're a vocal minority or if most subscribers to the -announce list would appreciate seeing such messages. Mark Michelson ___ -- 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] Dial weirdness
Bruce Ferrell wrote:
I'm seeing this response to SIP calls originated in the following manner:
Dial(SIP/${EXTEN}SIP/{$DID},30,r)
handle_response_invite: Re-invite to non-existing call leg on other UA.
The response is from the second part of the dial. What exactly does it
mean and how can I fix it?
Thanks in advance
Bruce
First of all, it may just be a transcription error on your part, but the
variable in the second part of the Dial statement should be ${DID} instead of
{$DID}.
That message you see on the console probably means that Asterisk has received a
481 response to the INVITE it has sent out. Apparently, whoever is receiving
the
INVITE thinks that it is a re-INVITE that belongs to an established SIP dialog,
but then it can't actually find the dialog to which the INVITE belongs. This
seems like it is an incorrect interpretation by the remote end since Asterisk
generates a new callid, new from-tag, and has no to-tag on each initial INVITE
it sends out when starting a call. It may be helpful to look at a packet
capture
from a failed attempt. It may be that whoever is sending back the 481 is
sending
a reason for it, or it may be that there is something obviously malformed in
the
SIP requests being sent by Asterisk.
Mark Michelson
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Re: [asterisk-users] Problems With Playback of Audio On SIP Only System
Kevin P. Fleming wrote: Mark Michelson wrote: If you are using gsm prompts and gcc version 4.2 or higher, then you may be experiencing the optimizer bug that gcc has with gsm audio. The workarounds for this are to use a different format for sounds or to set the DONT_OPTIMIZE flag in menuselect. If you want an optimized build and gsm formatted sounds, then you could always attempt downgrading your gcc version to 4.1 or earlier. This is affecting users frequently enough that we probably need to engineer some sort of configure-script test to check for this problem at build time. I started thinking about this and I'm not sure how you can check for this at configure-time or build-time. While it would be easy to check what gcc version is being used, it is not so easy (dare I say, not possible) to see if gsm-formatted sounds are going to be used in this setup. Besides the fact that core sounds are not placed on the system until make install is run, we can't know if there are out-of-tree gsm-formatted sound files which will be used, too. Did you have something clever in mind for such a check? Mark Michelson ___ -- 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 Playback of Audio On SIP Only System
Brian Alexander wrote: Mark, Thanks - that was the problem I was having. Is there somewhere I could have looked to have discovered the problem on my own? I would never have guessed that on my own and my searches had not found it either. Thanks again, -Brian In this particular case, I know of a bug report being reported on the Asterisk bug tracker (it has since been closed, though). I also think that it has been discussed on this mailing list before, too. The thing that makes it difficult to track is the fact that to you, it just sounded like garbled audio, so that's probably what you searched for. There have probably been hundreds of threads on that subject on this list, so filtering through it all is not easy. Mark Michelson ___ -- 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 Playback of Audio On SIP Only System
Brian Alexander wrote: I have been installing Asterisk as a SIP only system (no Digium Hardware) for demonstration purposes. SIP users can connect to menus and voicemail fine but the audio quality is terrible. The stock voicemail problems are bad but basically understandable - voice menus recorded through the asterisk-gui-2.0 are difficult to even understand. The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have configured the phone for ulaw to be it primary codec and set disallow all and allow ulaw in the users.conf. When that did not work I guessed that something was wrong with dahdi_dummy but dahdi_test is showing results around 99.987%. Here are the details of what software I have been using: asterisk-1.4 (r168975) dahdi-linux-complete 2.1.0 (r 5662) asterisk-gui-2.0 (r4446) The linux kernel is 2.6.24.6 built with 1000 Hz timer. Thank you for your help, I am a stumped. -Brian If you are using gsm prompts and gcc version 4.2 or higher, then you may be experiencing the optimizer bug that gcc has with gsm audio. The workarounds for this are to use a different format for sounds or to set the DONT_OPTIMIZE flag in menuselect. If you want an optimized build and gsm formatted sounds, then you could always attempt downgrading your gcc version to 4.1 or earlier. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote RTP
Gabriel Ortiz Lour wrote: Hi all, Suposing that 2 SIP phone register at a remote (internet) asterisk, what is the best way, if any, to make the RTP traffic go phone to phone, whithout using the internet conection (asterisk)? Thanks, Gabriel By default, Asterisk will attempt to offload the media from the server so that it may flow directly between the phones. There are several factors which may prevent this, though. For instance, if Asterisk is recording the call or needs to listen for DTMF in order to activate a specific feature, then Asterisk has to have the RTP flow through it. Mark Michelson ___ -- 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] bug 14153 and svn checkout.
Philipp Kempgen wrote: Jerry Geis schrieb: Jerry Geis wrote: I just did an SVN check out and the fix for bug 14153 was not included in the SVN checkout. Is there something special I need to issue in the SVN checkout to get it? I did not include the command I used. svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk using this command and looking at channels/chan_alsa.c the fixes are not included. Looks like http://bugs.digium.com/view.php?id=14153 was closed by accident (typo in the commit message). I think tilghman meant to close http://bugs.digium.com/view.php?id=14151 http://bugs.digium.com/view.php?id=14153 needs to be reopened. Philipp Kempgen I re-opened this bug. Thanks for bringing this up. Mark Michelson ___ -- 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, SIP channel and In Use
Benoit wrote: Hi, I'm a little surprised, up until 1.4.22 my agents where using an IAX channel to ZoIPer Softphone, however since after the upgrade to .22 we experienced a problem with hangup failure between zoiper and asterisk (look like bug http://bugs.digium.com/view.php?id=13184) i made them switch to SIP instead. Weird thing is that the 'Not In Use' warning message keep showing (WARNING[1863]: app_queue.c:3136 try_calling: The device state of this queue member, Agent/136, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings.) However, when i look at the queue a few sec after the Agent is marked 'in use' which wasn't the case with IAX iirc Agent are defined using a Local channel, but we used the '/n' flag to passthru the status: Agent/136 (Local/1...@queues/n) .. As for the SIP peer definition i have the limitonpeer=yes in the general section and all peers are templated based on this: [poste](!) type=friend host=dynamic qualify=yes call-limit=6 Is their something more in can do to avoid the warning ? I believe the use of the Local channel is what is causing the warning to appear. The problem is that the device state is not updated until after app_queue checks to see if it should display that warning. This has been brought up before, but since it doesn't actually adversely affect the operation of the queue, not much has been done. The worst you have to worry about is that warning. If you find the warning to be annoying, the best I can offer you is to either not log warnings (a bad idea, imho) or just remove that line of code from the source. Mark Michelson ___ -- 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 listen in on a SIP channel?
Ron McCarthy wrote: Hi list, I see their is ExtenSpy(), I want to monitor calls (in and out I hope) from another phone, all the channels are SIP. ChanSpy() looks like you cannot give it a context and I want to be able to only monitor certain calls. Any Ideals on how to do this? Thanks! You can use ChanSpy for this, using its grouping feature. When a call is made to or from a phone which you would like to listen to, set the SPYGROUP variable to some number. Then when you call the Chanspy application, supply it with the g option and use that number as an argument. You can get more details on this by issuing the command core show application Chanspy in the Asterisk CLI. Specifically look at the g option. Hope this helps. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue
Mateusz Pawlowski wrote: Hi, I was asked to create a Queue which instead of playing MoH it generates the ringing tone. I had a look around but could find anything, I would welcome and help. Regards Mateusz You can pass the 'r' option to the Queue application for this purpose. As an example: exten = 5000,1,Queue(MyQueue,r) Note that if you are using an Asterisk version prior to 1.6.0, this will have the side-effect of not playing any sort of configured sounds to the caller while he is waiting, e.g. hold time or position announcements. He will hear nothing but ringing until someone answers. Mark! ___ -- 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] Digium sites down for maintenance
Hi all, Sorry for the inconvenience but the following Digium-hosted sites are currently down for maintenance purposes: svn.digium.com svncommunity.digium.com bugs.digium.com packages.digium.com reviewboard.digium.com Apologies for the unannounced downtime; we will let you know when they are back up. Happy Holidays, Mark Michelson ___ -- 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] Join empty queue property
equis software wrote: I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. Unless you are using the trunk version of Asterisk right now, this can't be done very easily. In trunk, the joinempty setting for queues has been modified to not only allow the current allowed values of no, yes, loose, and strict, but also to allow a comma-separated list of conditions under which you consider a queue member to be unavailable. There is a detailed explanation of all the allowed conditions in the queues.conf.sample file in trunk's configs/ directory in the source. Since this feature is already in trunk but not in any released version of Asterisk, it will be present in Asterisk version 1.6.2. Mark Michelson ___ -- 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] Most Digium services are back on-line
Hello again, The following Digium-hosted services are no longer down and should be functioning properly: svn.digium.com svncommunity.digium.com packages.digium.com reviewboard.digium.com At this time, bugs.digium.com remains down, but we will have it back up as soon as we can. Thank you for your patience. Mark Michelson ___ -- 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] Directory exists when * is pressed....but where?
Mike wrote: I have been trying to figure out how the * works when in the Directory (dial-by-name). When I press * (which is supposed to exit the directory) I end up somewhere which I never specified. It seems like Asterisk just picked that place to go, because I never specified it. The wiki is no help on that one… Mike If you look at the help text for the Directory application using the Asterisk CLI (core show application directory), it specifies that pressing the '*' key will send you to the 'a' extension if it exists. Mark Michelson ___ -- 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] ael queue gosub already has PBX structure??
Giedrius Augys wrote:
Hello,
I want that after client and queue member call would be established,
cmd queue runs some 'procedures' . So I am using cmd Queue option
'gosub'. This is my example of ael :
context QUEUE {
_X. = {
Ringing();
Wait(4);
Answer();
Queue(${Queue},wr,,,60,,,check-record);
Hangup();
};
};
macro check-record() {
Set(MEMBERNUMBER=${CUT(MEMBERINTERFACE,@,1)});
Set(MEMBERNUMBER=${CUT(MEMBERNUMBER,/,2)});
return;
};
Everything works normal, but when the client's and queue call
establishes , I get this WARNING:
-- Local/1...@cc-out-da9a;1 answered SIP/xxx.xxx.xx-12d132d0
[Dec 17 20:52:12] WARNING[3849]: pbx.c:3656 __ast_pbx_run:
SIP/sip.call.lt-12d132d0 already has PBX structure??
== Starting SIP/sip.call.lt-12d132d0 at check-record,s,0 failed so
falling back to exten 's'
-- Executing [...@check-record:1] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=Local/123) in new stack
-- Executing [...@check-record:2] Set(SIP/sip.call.lt-12d132d0,
MEMBERNUMBER=123) in new stack
What I'm missing? Something wrong with ael syntax/structure ?
Thanks in advance
--
Pagarbiai / Best Regards,
Giedrius Augys
This is a bug you are experiencing, which I fixed recently in a series of
commits. Assuming you are using a 1.6 tag, the next build should have this
problem fixed.
Mark Michelson
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Re: [asterisk-users] Queue Question
Sebastian wrote: Is this going to be released in any 1.6 version soon?? Your branch (queue-reset) is supouse to be the same as trunk but with this functionality? Is this branch updated every time trunk is committed?? I checked the log and seems to have the latest commits of trunk, but I would like to be sure. Thanks Regards, Sebastian This branch is based off of Asterisk trunk and is automatically updated once an hour with the latest updates to trunk. As far as when this will make it into 1.6, it is unknown. There are still a few minor tweaks that need to be made and I'll probably do some testing and code review. Then it will need to go through the peer review process and a version will be set for its release. If I had to make an estimate, it will be in 1.6.2 at the earliest, but 1.6.3 seems more likely given that I haven't put a lot of work into this branch lately due to more pressing matters. Mark Michelson ___ -- 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] tcpdum
michel freiha wrote: Dear Sir, What I'm interested to is to know how much time the rtp packets takes from the time it access the asterisk server,to when it'll leave Is this function or variable exist anywhere? If you want statistics on RTP packets, then you should look into RTCP reporting. A simple facility for looking at this information would be the Asterisk CLI commands rtcp stats on and rtcp debug assuming that you are running Asterisk 1.4. If you are using Asterisk trunk, the commands are rtcp set stats on and rtcp set debug on. You may also be able to filter the RTCP packets in a program like wireshark and analyze them there as well. Mark Michelson ___ -- 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 Question
Sebastian wrote: Hi, In queues realtime, when the queue start and when it ends. I mean, for example to calculate service level, how many calls, etc. If I want to start the queue from with 0 calls, etc, how do I do this? And if I want to stop it, so I can start it again?? Thanks!! Regards, Sebastian Currently, there is not a way to do this with realtime queues. During a reload, realtime queues are not touched at all. I have a development branch set up which is supposed to help this as well as other rigidities present when it comes to reloading and resetting queues. The branch is located at the following URL if you wish to give it a test: http://svn.digium.com/svn/asterisk/team/mmichelson/queue-reset If you run the code there, you'll find that there is a command called queue reset stats which should do what you want. Mark Michelson ___ -- 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] MSet()
Philipp Kempgen wrote: How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? Philipp Kempgen It is a Multiset application. My recollection of the addition is that due to parser changes in 1.6, a statement like: exten = s,1,Set(FOO=hello,BAR=world) would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. Mark Michelson ___ -- 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] MSet()
Mark Michelson wrote: Philipp Kempgen wrote: How is MSet() different from Set()? Is it supposed to be a Multi-Set()? Why was it added in 1.6? Philipp Kempgen It is a Multiset application. My recollection of the addition is that due to parser changes in 1.6, a statement like: exten = s,1,Set(FOO=hello,BAR=world) would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. Mark Michelson An even better answer is in the UPGRADE-1.6.txt document in the Asterisk source: * The behavior of the Set application now depends upon a compatibility option, set in asterisk.conf. To use the old 1.4 behavior, which allowed Set to take multiple key/value pairs, set app_set=1.4 in [compat] in asterisk.conf. To use the new behavior, which permits variables to be set with embedded commas, set app_set=1.6 in [compat] in asterisk.conf. Note that you can have both behaviors at the same time, if you switch to using MSet if you want the old behavior. Mark Michelson ___ -- 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] Gosubs broken since r160626 (1.6.0 SVN) ?
Gary Hawkins wrote:
Hi all,
I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs
have stopped working.
This is from the verbose logs:
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/aaisp-3802,
1?5:7) in new stack
-- Goto (incoming-aaisp,0407271,5)
-- Executing [EMAIL PROTECTED]:5] Gosub(IAX2/aaisp-3802,
macro-announcement,s,1(anonymous_call_rejection,22)) in new stack
== Spawn extension (incoming-aaisp, 0407271, 6) exited non-zero on
'IAX2/aaisp-3802'
-- Hungup 'IAX2/aaisp-3802'
This was the original AEL2 code:
0407271 = {
Verbose(We got here);
AGI(caller_id_rewriter/caller_id_rewriter.py);
Set(CALLERID(name)=1 ${CALLERID(name)});
if (${WITHHELD} = yes) {
macro-announcement(anonymous_call_rejection,22);
Hangup(22);
}
Dial(${ALLPHONES},20);
if (${DIALSTATUS} = BUSY) {
VoiceMail(201,b);
}
else
{
VoiceMail(201,u);
}
Hangup(${HANGUPCAUSE});
}
This was working on 1.6.0 SVN before r160626 and I have not changed any
of the code. The Gosubs were generated by the AEL parser. In the AEL2
dialplan I am calling
macro-announcement(anonymous_call_rejection,22);
Has anyone seen similar problems to this?
Thanks
Gary H
This appears to be a side-effect of a bug fix I made.
To give some background, one of the changes in that commit was to check for the
existence of the extension which you are trying to go to to execute the gosub.
If it does not exist, then we back out with an error.
Looking at the console output, there appears to be a problem with the AEL
parser. It is attempting to send you to the macro-announcement context,
extension s, priority 1, with label anonymous_call_rejection,22. I assume that
anonymous_call_rejection and 22 were supposed to be arguments to the gosub, and
not treated as a label. Since no extension exists with that label, that is why
the gosub is now failing.
This is definitely a bug and needs to be corrected before the next version of
1.6.0 is released.
Mark Michelson
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Re: [asterisk-users] AMI interface problem
Jim Dickenson wrote: I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel: SIP/GXP280-18 Application: queue data: tqe ActionID: callandqueue Callerid: 12 Async: true And I get this response: Response: Error ActionID: callandqueue Message: Channel not specified Is anyone else seeing anything like this? Thanks for pointing this out. I have located the erroneous code and have fixed it in subversion, revision 161490. The next rc of 1.6.0 will not have this bug. Mark Michelson ___ -- 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] Gosubs broken since r160626 (1.6.0 SVN) ?
Mark Michelson wrote:
Gary Hawkins wrote:
Hi all,
I've just upgraded to latest 1.6.0 SVN from a few days ago and my Gosubs
have stopped working.
This is from the verbose logs:
-- Executing [EMAIL PROTECTED]:4] GotoIf(IAX2/aaisp-3802,
1?5:7) in new stack
-- Goto (incoming-aaisp,0407271,5)
-- Executing [EMAIL PROTECTED]:5] Gosub(IAX2/aaisp-3802,
macro-announcement,s,1(anonymous_call_rejection,22)) in new stack
== Spawn extension (incoming-aaisp, 0407271, 6) exited non-zero on
'IAX2/aaisp-3802'
-- Hungup 'IAX2/aaisp-3802'
This was the original AEL2 code:
0407271 = {
Verbose(We got here);
AGI(caller_id_rewriter/caller_id_rewriter.py);
Set(CALLERID(name)=1 ${CALLERID(name)});
if (${WITHHELD} = yes) {
macro-announcement(anonymous_call_rejection,22);
Hangup(22);
}
Dial(${ALLPHONES},20);
if (${DIALSTATUS} = BUSY) {
VoiceMail(201,b);
}
else
{
VoiceMail(201,u);
}
Hangup(${HANGUPCAUSE});
}
This was working on 1.6.0 SVN before r160626 and I have not changed any
of the code. The Gosubs were generated by the AEL parser. In the AEL2
dialplan I am calling
macro-announcement(anonymous_call_rejection,22);
Has anyone seen similar problems to this?
Thanks
Gary H
This appears to be a side-effect of a bug fix I made.
To give some background, one of the changes in that commit was to check for
the
existence of the extension which you are trying to go to to execute the
gosub.
If it does not exist, then we back out with an error.
Looking at the console output, there appears to be a problem with the AEL
parser. It is attempting to send you to the macro-announcement context,
extension s, priority 1, with label anonymous_call_rejection,22. I assume
that
anonymous_call_rejection and 22 were supposed to be arguments to the gosub,
and
not treated as a label. Since no extension exists with that label, that is
why
the gosub is now failing.
This is definitely a bug and needs to be corrected before the next version of
1.6.0 is released.
Mark Michelson
In a fit of wild curiosity, I decided to double-check to be sure that the
problem was an AEL parser issue and not one of my own. I actually discovered a
bug introduced by my changes. I have fixed this bug in revision 161494 of the
1.6.0 branch. I suspect this will fix the problem you were seeing, too.
Mark Michelson
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Re: [asterisk-users] Dahdi and ztdummy
Mike wrote: Hi, I need to run ztdummy for Paging, but now that this is all become dahdi I don`t really know where to start. I did build dahdi before building asterisk, but that`s it. I find it hard to find any documentation referring to dadhi instead of zaptel. I have no Digium hardware, but I still need the ztdummy timer (or whatever it`s called now). How do I get myself going? Regards,** * * *Mike* DAHDI has 'dahdi_dummy' in place of ztdummy. You should be able to use it exactly the same way that you used ztdummy. Mark Michelson ___ -- 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
