Re: [asterisk-users] Freepbx / Asterisk PJsip multipe devices
Eric sent you a link of how to do it at the PJSIP level. To do it in the dial plan, something like exten => _6XXX,1,Dial(PJSIP/SoftPhone/HardwarePhone) Mitch On 2/6/19 8:32 AM, basti wrote: that was my first idea. and how should an other user know which number he should dial? user a: soft phone extension 100 hardware phone extension 101 On 06.02.19 15:25, Mitch Claborn wrote: You can do this in the dial plan. Register the devices separately and include both addresses in the Dial() command. Mitch On 2/6/19 8:16 AM, basti wrote: In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: both phones are in the same net. when the soft phone is shut down, on hardware phone only an led is flashing to show an incoming call but no sound. both phones use the same extension. that is the reason why I use pjsip. On 06.02.19 14:59, Antony Stone wrote: On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: These two phones are not using the same extension, are they? If you shut down the softphone, does the hardware phone then ring? Antony. On 2/6/2019 8:49 AM, basti wrote: both phones are registered. and the hardware phone can also make calls. but an incoming call is not displayed and also not hearing. Call Waiting is also disabled. On 06.02.19 14:07, Cyril Alberts wrote: Hi, look at your registrations, is the hardware phone registered? if yes, which phone vendor do you want to connect? can you make outgoing calls with hardwarephone? BR Cyril Am Mittwoch, den 06.02.2019, 13:00 +0100 schrieb basti: Hello, I have some user that had have a hardwarephone and an softphone. I use pjsip driver and set "Max Contacts = 2" to have register both at the same time. But Only the softphone is ring. the hardware phone is mute. How can i fix this? -- _ -- 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] Freepbx / Asterisk PJsip multipe devices
You can do this in the dial plan. Register the devices separately and include both addresses in the Dial() command. Mitch On 2/6/19 8:16 AM, basti wrote: In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: both phones are in the same net. when the soft phone is shut down, on hardware phone only an led is flashing to show an incoming call but no sound. both phones use the same extension. that is the reason why I use pjsip. On 06.02.19 14:59, Antony Stone wrote: On Wednesday 06 February 2019 at 13:54:44, Mark Wiater wrote: These two phones are not using the same extension, are they? If you shut down the softphone, does the hardware phone then ring? Antony. On 2/6/2019 8:49 AM, basti wrote: both phones are registered. and the hardware phone can also make calls. but an incoming call is not displayed and also not hearing. Call Waiting is also disabled. On 06.02.19 14:07, Cyril Alberts wrote: Hi, look at your registrations, is the hardware phone registered? if yes, which phone vendor do you want to connect? can you make outgoing calls with hardwarephone? BR Cyril Am Mittwoch, den 06.02.2019, 13:00 +0100 schrieb basti: Hello, I have some user that had have a hardwarephone and an softphone. I use pjsip driver and set "Max Contacts = 2" to have register both at the same time. But Only the softphone is ring. the hardware phone is mute. How can i fix this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] features.conf disconnect and local channels
Asterisk 16.1
This statement appears in the features.conf doc: "Note that the DTMF
features listed below only work when two channels have answered and are
bridged together. They can not be used while the remote party is ringing
or in progress. If you require this feature you can use chan_local in
combination with Answer to accomplish it."
I need attended transfer and disconnect from features.conf to work.
Below is what I came up with that seems to work fine. Is there a better
way? This seems a bit verbose.
[InternalSets]
exten =>298,1,NoOp()
same =>n,Dial(Local/M${EXTEN}@InternalSets/nj,,H)
exten =>M298,1,NoOp()
same =>n,Answer()
same =>n,GoSub(sub-voicemail,start,1(${MITCHIPHONE},${EXTEN:1}))
exten =>299,1,NoOp()
same =>n,Dial(Local/M${EXTEN}@InternalSets/nj,,H)
exten =>M299,1,NoOp()
same =>n,Answer()
same =>n,GoSub(sub-voicemail,start,1(${MLCX450},${EXTEN:1}))
[sub-voicemail]
do some checks and then Dial or send to voicemail.
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Re: [asterisk-users] Overhead pager announcement in "background" channel
Here's how I solved this: I use the System() dialplan application to
call out to a bash script, which creates asterisk call files and moves
them to the proper directory. I opted to send notification to a select
group of staff members rather than using the overhead pager, but the
same technique would work for the pager.
The script is very fast and does not interrupt the flow of the actual call.
Mitch
On 1/12/19 8:57 PM, Mitch Claborn wrote:
We have an overhead paging system that is working fine with our asterisk
16.1 server. I'd like to be able to push an announcement to the paging
extension (PJSIP) without disrupting the current channel. Can this be
done? I want to use it in the dial plan when a 911/emergency call is
placed, so it is imperative that the calling channel not be disrupted.
The following works, but it disrupts the calling channel:
Dial(${PAGER},20,A(filename)L(3000))
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[asterisk-users] Outbound caller ID ignored
Setting the outbound caller ID works fine on our PRI (T1) lines, but does not work on our local POTS lines. No errors in the logs, the new caller ID just seems to be ignored. Should I expect it to work on the analog lines? Dial plan: same =>n,Set(CALLERID(all)=111222) same =>n(dialit),Dial(DAHDI/50/1222333,30) Channels: signalling=fxs_ks callerid=asreceived group=20,21 context=from-pstn faxdetect=incoming faxdetect_timeout=0 faxbuffers => 12,half channel => 49-53 -- Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Overhead pager announcement in "background" channel
We have an overhead paging system that is working fine with our asterisk
16.1 server. I'd like to be able to push an announcement to the paging
extension (PJSIP) without disrupting the current channel. Can this be
done? I want to use it in the dial plan when a 911/emergency call is
placed, so it is imperative that the calling channel not be disrupted.
The following works, but it disrupts the calling channel:
Dial(${PAGER},20,A(filename)L(3000))
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Mitch
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[asterisk-users] Timeout for AGI/HAGI connections
Asterisk 16.1.0 I'm using hagi and SRV records for a "high availability" configuration of AGI servers. When the first AGI server in the list is completely down, asterisk quickly moves on to the next one. That is all good. My concern is what will happen if asterisk can actually connect to the first AGI server and initiate the script, but something is internally wrong with the server and it takes a long time to respond. Is there some way to set a timeout value, so that if the AGI server/script does not respond in (some amount of time) that asterisk will time out and treat it as a failure? Even better would be if that timeout would trigger a retry on the next server in the SRV record list. -- Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DPMA - simulate mDNS scan from command line
I'm working on an asterisk upgrade to 16.1 and am remote from that location. We use Digium phones there, configured with DPMA. From my VPN I can connect to the server directly with the phone on my desk, but it doesn't find the configuration server automatically since I'm on a different physical network. Is there a way to simulate on the linux command line whatever mDNS scan the phone does when looking for a configuration server, so that I can verify that it is set up correctly? Answering my own question: avahi-browse --resolve _digiumproxy._udp That seems to do the trick. -- Mitch -- _ -- 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] Asterisk 16.1.0 Now Available
When building a new release, is it possible to copy the output of "make menuselect" from a previous build directory? If so, what files need to be copied? That would save some time in the upgrade process. Mitch On 12/11/18 4:11 PM, Asterisk Development Team wrote: The Asterisk Development Team would like to announce the release of Asterisk 16.1.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk -- _ -- 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] DAHDI fax detection
Thanks Ryan. Would you mind sharing snippets of your DAHDI channel config and dialpaln? Mitch On 12/11/18 8:43 AM, Ryan, Travis wrote: Yes it's very easy. Mine is using a simulated PRI over an ATT Flex line. I just followed the many tutorials out there. I answer the call, then it takes 6-7 seconds (you can add a wait if you want) and then it detects it and drops it to the fax extension in the same context. Also, until recently I used the Asterisk Fax licensing, but have since switched to spandsp as that is the supported one ongoing. -- _ -- 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] DAHDI fax detection
I'm assuming that no one knows the answer to this. Does anyone have fax detection successfully working? If so, can you share your configuration? Mitch On 12/4/18 4:27 PM, Mitch Claborn wrote: Asterisk 16 latest DAHDI 3.0.0 latest Excerpt from chan_dahdi.conf is shown below. I'm trying to enable fax detection on inbound calls so that I can take appropriate action in the dial plan. "dahdi show channel 1" shows "Fax Handled: no". Does that mean that I don't have it configured correctly? [channels] ; Span 1: WCTE2/0/1 "WCTE23X (PCI) Card 0 Span 1" (MASTER) ESF/B8ZS RED group=10,11 context=from-pstn switchtype = national signalling = pri_cpe faxdetect=incoming faxdetect_timeout=0 faxbuffers => 12,half channel => 1-23 ; Span 2: WCTE2/0/2 "WCTE23X (PCI) Card 0 Span 2" ESF/B8ZS RED group=10,12 context=from-pstn switchtype = national signalling = pri_cpe channel => 25-47 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI fax detection
Asterisk 16 latest DAHDI 3.0.0 latest Excerpt from chan_dahdi.conf is shown below. I'm trying to enable fax detection on inbound calls so that I can take appropriate action in the dial plan. "dahdi show channel 1" shows "Fax Handled: no". Does that mean that I don't have it configured correctly? [channels] ; Span 1: WCTE2/0/1 "WCTE23X (PCI) Card 0 Span 1" (MASTER) ESF/B8ZS RED group=10,11 context=from-pstn switchtype = national signalling = pri_cpe faxdetect=incoming faxdetect_timeout=0 faxbuffers => 12,half channel => 1-23 ; Span 2: WCTE2/0/2 "WCTE23X (PCI) Card 0 Span 2" ESF/B8ZS RED group=10,12 context=from-pstn switchtype = national signalling = pri_cpe channel => 25-47 -- Mitch -- _ -- 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] asterisk is not seeing my queues in database
Maybe post the result from that query here? Mitch On 12/4/18 10:46 AM, Dominic wrote: I enabled the logs on the mysql database and ran : realtime load queues name cou0002-test in the mysql log I can see that the proper select statement is being executed: 2018-12-04T16:29:27.253094Z 229 Query SET SESSION TRANSACTION ISOLATION LEVEL READ COMMITTED 2018-12-04T16:29:27.254384Z 229 Prepare SELECT * FROM queues WHERE name = ? 2018-12-04T16:29:27.254902Z 229 Execute SELECT * FROM queues WHERE name = 'cou0002-test' 2018-12-04T16:29:27.255606Z 229 Close stmt I also ran the query (SELECT * FROM queues WHERE name = 'cou0002-test') on the db and I do get a result. On Tue, Dec 4, 2018 at 9:08 AM Mitch Claborn <mailto:mitch...@claborn.net>> wrote: Maybe try capturing the queries that are executed on the mysql server? That might point you in the right direction. -- show the log file name SHOW VARIABLES LIKE 'general_log%'; -- turn logging on and off SET GLOBAL general_log='ON'; SET GLOBAL general_log='OFF'; Mitch On 12/4/18 7:50 AM, Dominic wrote: > Hi I am facing an issue where asterisk cannot see the queues that exist > in my database through realtime. I am using res_odbc and a local mysql > database. > > If I run: > > realtime load queues name myqueue > > I get "No rows found matching search criteria.", however if I do the > same for a peer: > > realtime load sippeers name > > Then I get a result. Since my queues table is in the same database as my > sippeers table, I was expecting consistent result between the two. > > I am a bit stuck here on where to look for errors or how I can debug > this issue, I can't see any error messages when I call the Queue > application besides "queue_exec: Unable to join queue". Also, this is an > almost exact copy of an existing Asterisk, so I'm confident the table > structure is correct but I'm obviously missing something. > > Any suggestions? > -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connected line update prevented
I am seeing the following type of error in the console and verbose log.
Connected line update to PJSIP/mlc296- prevented
It is happening after a Dial command [Dial("PJSIP/mlc296-0006",
"PJSIP/mlcx450,25,IktT")] before the other party answers the phone.
This happens to be dialing from a Digium phone to a soft phone, but I
also get the message when dialing the other way.
I am using the latest Asterisk 16 and DPMA.
There doesn't seem to be any damage - everything works OK, but I'd like
to figure out what this means and fix it or prevent it.
I did find
https://wiki.asterisk.org/wiki/display/DIGIUM/Digium+Phones+and+Connected+Line+Updates
and applied those changes but it did not help.
Any ideas?
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Re: [asterisk-users] asterisk is not seeing my queues in database
Maybe try capturing the queries that are executed on the mysql server? That might point you in the right direction. -- show the log file name SHOW VARIABLES LIKE 'general_log%'; -- turn logging on and off SET GLOBAL general_log='ON'; SET GLOBAL general_log='OFF'; Mitch On 12/4/18 7:50 AM, Dominic wrote: Hi I am facing an issue where asterisk cannot see the queues that exist in my database through realtime. I am using res_odbc and a local mysql database. If I run: realtime load queues name myqueue I get "No rows found matching search criteria.", however if I do the same for a peer: realtime load sippeers name Then I get a result. Since my queues table is in the same database as my sippeers table, I was expecting consistent result between the two. I am a bit stuck here on where to look for errors or how I can debug this issue, I can't see any error messages when I call the Queue application besides "queue_exec: Unable to join queue". Also, this is an almost exact copy of an existing Asterisk, so I'm confident the table structure is correct but I'm obviously missing something. Any suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Understanding local channels
Can someone point me to a good tutorial / explanation of local channels? I've been using them without really understanding what is going on, and we all know how dangerous that is! I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels but I'm just not quite getting it. -- Mitch -- _ -- 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] Understanding local channels
Here's my current specific scenario. I have a working call me now
solution on our web site. The customer types in their phone number, it
goes into our normal sales asterisk queue via an AMI action. When the
agent answers the call, he gets a brief announcement then asterisk dials
the customer's number. (This works in Asterisk 11. There is an
apparent bug in asterisk 12 with queue variables:
https://issues.asterisk.org/jira/browse/ASTERISK-24267) It works, but
I'm struggling to understand how.
*AMI Action:*
Action: Originate
Channel: Local/s@callmenow/n
Context: dial-to-customer
Exten: s
Priority: 1
Async: true
Variable: MMCALLMENOWID=107
Timeout: 99
Callerid: Call Me Now 778
*Dial Plan:*
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,Rtc)
[dial-to-customer]
exten = s,1,NoOp(dial-to-customer channel=${CHANNEL(name)})
same =n,Wait(1)
same =n,Playback(custom/callmenow-announce)
; do some more stuff
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,TKU(dial-to-cust-connect-sub))
Mitch
On 08/25/2014 11:43 AM, Joshua Colp wrote:
On 8/25/2014 1:33 PM, Patrick Laimbock wrote:
On 25-08-14 17:06, Mitch Claborn wrote:
Can someone point me to a good tutorial / explanation of local
channels? I've been using them without really understanding what is
going on, and we all know how dangerous that is!
I've read http://www.voip-info.org/wiki/view/Asterisk+local+channels
but
I'm just not quite getting it.
How about the info on the Asterisk wiki:
https://wiki.asterisk.org/wiki/displa/AST/Introduction+to+Local+Channels
That wiki page isn't REALLY detailed. To what level are you wanting to
know more about, Mitch?
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[asterisk-users] Asterisk 12 - queue variables not passed to local channel
Asterisk 12.5
I'm using AMI to initiate a call me now feature from the web site.
The AMI looks like:
Action: Originate
Channel: Local/s@callmenow
Context: dial-to-customer
Exten: s
Priority: 1
Async: true
Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/111222
Timeout: 99
Dial Plan:
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,Rtc)
[dial-to-customer]
exten = s,1,NoOp(dial-to-customer channel=${CHANNEL(name)})
same =n,DumpChan()
The dial-to-customer context is invoked when the sales queue agent
answers the phone.
When the local channel is used, the queue related variables,
specifically MEMBERINTERFACE, are missing. When a normal call
(typically SIP or DAHDI channel) enters the queue, the MEMBERINTERFACE
and other variables are present.
my queues.conf has
setinterfacevar = yes
setqueueentryvar = yes
setqueuevar = yes ;
I didn't see anything in the V12 doc that related to this.
Is this a bug or a feature?
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[asterisk-users] AMI CoreShowChannel missing Application field
Asterisk 12.5 The CoreShowChannel event (in response to the CoreShowChannels action) no longer returns the Application field as it did in Asterisk 11. Is this a bug or a feature? -- Mitch -- _ -- 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] AMI CoreShowChannel missing Application field
On 08/22/2014 02:47 PM, Matthew Jordan wrote: Yup, that's a bug. When things got ported over to hit the cached snapshots of the channels (as opposed to locking the live channel), that field got missed. Please file a bug on issues.asterisk.org. Thanks! Matt https://issues.asterisk.org/jira/browse/ASTERISK-24262 -- _ -- 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] diagnostic info for a segfault
Asterisk 12.5 I have a reproducible segfault using the MeetMe application. How do I gather the necessary information (backtrace, core dump...) to submit a bug report? -- Mitch -- _ -- 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] DPMA: User SIP settings missing or invalid
Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit [2014-08-21 16:37:49] WARNING[5797]: phone_users.c:5236 set_and_process: User SIP settings missing or invalid I'm getting the error message above when DPMA is enabled, using a fresh build but with my config files from Asterisk 11. Any idea what it means? I can't find the phone_users.c file to examine the source (assuming it is part of DPMA which has no source, rather than Asterisk). I can't tell that anything is not working - I can configure and connect at least one Digium phone. -- Mitch -- _ -- 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] DPMA: No provider found for label CustomPresence
Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit WARNING[5797]: presencestate.c:147 ast_presence_state_helper: No provider found for label CustomPresence ERROR[5797]: pbx.c:4375 ast_func_write: Function PRESENCE_STATE not registered I only see these when DPMA is enabled. Any ideas what causes this or how to correct it? -- Mitch -- _ -- 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] DPMA: No provider found for label CustomPresence
It appears to be loaded. This is a fresh build of Asterisk 12.5 from source. *CLI module show like func_presencestate.so Module Description Use Count Status func_presencestate.so Gets or sets a presence state in the dia 0 Running 1 modules loaded Mitch On 08/21/2014 06:55 PM, George Joseph wrote: Make sure the func_presencestate.so module is being loaded. Did you compile Asterisk yourself or are you using a pre-built from a distro? On Thu, Aug 21, 2014 at 5:34 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: Asterisk 12.5.0 DPMA 12.0_2.0.0 Ubuntu 12.04 64 bit WARNING[5797]: presencestate.c:147 ast_presence_state_helper: No provider found for label CustomPresence ERROR[5797]: pbx.c:4375 ast_func_write: Function PRESENCE_STATE not registered I only see these when DPMA is enabled. Any ideas what causes this or how to correct it? -- Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error opening file for reading: Permission denied
No, that's not it. The wording is different.
Mitch
On 08/18/2014 02:28 PM, Paul Greenberg wrote:
Mitch,
Is it the below error?
if ((fd = open(filename, O_RDONLY)) 0) {
ast_log(LOG_WARNING, Cannot open file '%s' for reading:
%s\n, filename, strerror(errno));
return NULL;
}
Regards,
Paul
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Mitch Claborn
mitch...@claborn.net
Sent: Monday, August 18, 2014 1:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Error opening file for reading: Permission denied
Asterisk 12.4
I am seeing message Error opening file for reading: Permission denied
several times during the asterisk startup (asterisk -cv) but it
doesn't say which file. Is there a way to find out which file is having
trouble?
--
Mitch
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Re: [asterisk-users] Error opening file for reading: Permission denied
I tried grep too. No 3rd party modules - this is an out-of-the box download and build. I'm guessing that some library function is being called to read a file and the error is happening there? Mitch On 08/19/2014 02:33 PM, Matthew Jordan wrote: On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net wrote: No, that's not it. The wording is different. grep doesn't turn up your phrase: ~/projects/12$ grep --include=*.c --include=*.h -r Error opening file . ~/projects/12$ Are you using any 3rd party modules that aren't delivered with Asterisk? -- _ -- 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] Error opening file for reading: Permission denied
Grepping the output of the strace revealed this: stat(/root/.terminfo, 0x7fff8622ed50) = -1 EACCES (Permission denied) open(/root/.asterisk_history, O_RDONLY) = -1 EACCES (Permission denied) open(/root/.odbcinst.ini, O_RDONLY) = -1 EACCES (Permission denied) [this one many times] That must be because I'm starting asterisk as root. When I su to asterisk first, then start it, those above disappear. Problem solved! Thanks Steve! Mitch On 08/19/2014 03:39 PM, Steve Edwards wrote: On Tue, Aug 19, 2014 at 11:36 AM, Mitch Claborn mitch...@claborn.net No, that's not it. The wording is different. Can you run Asterisk via strace? Something like: sudo -u asterisk strace /usr/sbin/asterisk -c -p -U asterisk -- _ -- 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] Error opening file for reading: Permission denied
Asterisk 12.4 I am seeing message Error opening file for reading: Permission denied several times during the asterisk startup (asterisk -cv) but it doesn't say which file. Is there a way to find out which file is having trouble? -- Mitch -- _ -- 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] Copying menuselect options
Is it possible (and advisable) to copy menuselect options from Asterisk 11 to Asterisk 12? If so, is menuselect.makeopts the only file to copy? -- Mitch -- _ -- 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] Asterisk 12 and DPMA
I read somewhere that DPMA is not supported for Asterisk 12. Can anyone confirm or deny that? If not supported yet, will it be? If so, when? -- Mitch -- _ -- 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] Notification when queue member's phone rings
Short question: how to get control or notification (gosub, macro, AGI) when a queue member's phone starts ringing due to an incoming call from the queue. Backround: Our phone operators serve both an asterisk call queue and a queue for web chat support. I have a gosub on the queue that calls to our app server to mark the operator unavailable for web chat as soon as they answer an incoming queue call. Similarly, when a web chat is connected, it uses AMI to tell asterisk to take the operator out of the phone queue. The other day, one operator got a web chat that came in while her phone was ringing with a queue call, so that neither remove from queue operation was effective in time. If I could get notification when the phone starts ringing I can reduce the window of opportunity for that by several seconds. It's only happened once in 2 years that I know of, so may not be worth worrying about. -- Mitch -- _ -- 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] Debugging stuck inbound call
Asterisk 11.1.0 running on Ubuntu 12.04 64 bit Dahdi Digium T1 card Occasionally, I will find an inbound call that just seems to be stuck, usually in our after-hours menu portion of the dial plan. This morning I had this one core show channels concise DAHDI/i1/5184097869-1baf2!MainMenu!s!20!Up!BackGround!custom/aa-night-hellocustom/hours_8:0-17:0_0:0-0:0_0:0-0:0custom/aa-night-instructions!5184097869!!!3!9393!(None)!sip1-1396004671.285644 which had been there for about 2.5 hours (time from core show channels verbose). The inbound channel here is to our toll free number on the T1. When we've researched these in the past, we've not found a correspondingly long call on the phone bill, leading me to wonder if the call is actually being disconnected, but Asterisk just doesn't find out. How can I go about debugging this? Are the dahdi commands that can show me the connection status from the hardware perspective? -- Mitch -- _ -- 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] calls processed value definition
The core show channels verbose command shows a calls processed value. Mine is currently 1928273. Exactly what does this figure represent? How is a call defined in this context? -- Mitch -- _ -- 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] Capture dead phone?
I certainly agree that the first and best solution is to deal with the hardware issues, and we've started working on that already. I'll investigate the suggested Asterisk ideas and post here if anything works for my purposes. Mitch On 11/08/2013 12:13 AM, Mikhail Lischuk wrote: Mitch Claborn писал 08.11.2013 02:51: Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Maybe you can connect agents and callers via MeetMe, and when AMI gets the MeetMe Leave event, put the caller on hold and return him to the queue (maybe in the first position). Just a guess, for I've never used such setup. But I strongly agree with people who say you'd better change your hardware. -- With Best Regards Mikhail Lischuk mailto:mlisc...@itx.com.ua -- _ -- 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] Capture dead phone?
Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will knock something loose with their foot. If/when that happens while the agent is on a call with a customer, I'd like to be able to save that caller and put them back in the queue (at the head of the queue). -- Mitch -- _ -- 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] Queues: Knowing when a caller is position 1 (agent phone ringing)
We do something very similar. Use the gosub parameter of the Queue application to call a subroutine in the dial plan when the agent answers the call. same =n,Queue(sales,tc,,sub-QueueConnected) [sub-QueueConnected] ; this runs on the agent/member's channel exten =s,1,NoOp() ; whatever you need to do here same =n,Return() See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue Mitch On 08/03/2013 12:45 PM, Timothy Smith wrote: Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this information either via netcat (to a client such as yac) to a Windows PC but the problem is I do not know when the caller is about to be connected to the agent, so that I run the command. If I wasn't using queues, it would be easy because I would run the netcat command and then dial the user's extension. My Question is: Is there a way I can know when the caller is just about to be connected to an agent (when the agent's SIP extension starts ringing)? There are these settings setinterfacevar, setqueueentryvar, setqueuevar in queues.conf but when can I use them? Have you guys been in this situation before? Any alternative solutions (sending caller info to an agent)? I am using Asterisk 11 and Windows 7 PCs for agents. Thank you! Kind Regards, Wilson -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial application b subroutine arguments not passing?
Asterisk 11.1.0
I'm trying to use the b subroutine of the Dial application so that I
can do some stuff with our internal applications that need to have
access to the called channel information. I can see that the subroutine
is being executed, but the arguments I pass don't see to make it to the
subroutine.
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,tc)
[dial-to-customer]
exten = s,1,NoOp(to-customer)
same =n,Wait(1)
same =n,Playback(custom/callmenow-announce)
same =n,GoSub(sub-outbound_caller_id,start,1)
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,*b(dial-to-customer-sub,s,1,${MMCUSTOMER_NUMBER},${MEMBERINTERFACE},${MEMBERNAME})*)
[dial-to-customer-sub]
; this runs on the customer's channel
exten =s,1,NoOp()
same =n,Set(OPERATORID=${ODBC_OPERATORID_FROM_ADDRESS(${ARG2})})
same =n,Verbose(2, dial-to-customer-sub interface ${ARG2} name
${ARG3} customer number ${ARG1} operatoriod ${OPERATORID} channel name
${CHANNEL(name)} unique ID ${CHANNEL(uniqueid)} )
same =n,Return()
The whole thing is kicked off by an AMI request:
Action: Originate
Channel: Local/s@callmenow
Context: dial-to-customer
Exten: s
Priority: 1
Async: true
Callerid: Call Me Now 777
Variable: MMCUSTOMER_NUMBER=9995551212
Timeout: 99
Output from the subroutine:
-- Executing [s@dial-to-customer-sub:3] Verbose(SIP/voipms-001e,
2, dial-to-customer-sub interface name customer number operatoriod
channel name SIP/voipms-001e unique ID mlcx500-1375465508.61 ) in
new stack
The U subroutine seems to work OK
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,U(dial-to-customer-sub,${MMCUSTOMER_NUMBER},${MEMBERINTERFACE},${MEMBERNAME}))
I want the b subroutine, because it is call before attempting to
connect the remote end. This gives me plenty of time to notify my
application and have it look up the customer's record while the call is
being placed. The U subroutine is called after the call is connected.
Am I missing something, or is it broke?
(This whole thing is development for a call me now feature from the
web site.)
--
Mitch
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Re: [asterisk-users] Dial application b subroutine arguments not passing?
On 08/02/2013 01:28 PM, Matthew Jordan wrote:
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Asterisk 11.1.0
I'm trying to use the b subroutine of the Dial application so that
I can do some stuff with our internal applications that need to have
access to the called channel information. I can see that the
subroutine is being executed, but the arguments I pass don't see to
make it to the subroutine.
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,tc)
[dial-to-customer]
exten = s,1,NoOp(to-customer)
same =n,Wait(1)
same =n,Playback(custom/callmenow-announce)
same =n,GoSub(sub-outbound_caller_id,start,1)
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,*b(dial-to-customer-sub,s,1,${MMCUSTOMER_NUMBER},${MEMBERINTERFACE},${MEMBERNAME})*)
Use a '^' to delineate arguments pass to subroutines. This is actually
true for the U option as well. See:
https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers
And:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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That is not working for me either.
same
=n,Dial(${TOLL}/${MMCUSTOMER_NUMBER},,b(dial-to-customer-sub^s^1^fred^$george^$arrrgh))
output is
-- Executing [s@dial-to-customer:8] Dial(SIP/mlcm800-0039,
SIP/voipms/9725232703,,b(dial-to-customer-sub^s^1^fred^$george^$arrrgh))
in new stack
-- SIP/voipms-003a Internal Gosub(dial-to-customer-sub,s,1) start
-- Executing [s@dial-to-customer-sub:2] Verbose(SIP/voipms-003a,
2, number interface name ) in new stack
PS - a link from the Dial page to the Pre-Dial Hanlders page would be
useful.
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Re: [asterisk-users] RED on DAHDI channel
I am running 2.6.1. I'll give the 2.6.y a try. Mitch On 05/28/2013 10:53 AM, Shaun Ruffell wrote: On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? Hi Mitch, What version of DAHDI are you using? Unfortunately I did insert a bug on 2.6.0 where it was possible for a channel on an AEX410 to get stuck in RED alarm depending on the timing from the central office. If you're not using 2.6.0+ you can ignore the remainder of this email. The bug was fixed in 2.6.2 in wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. [1] but unfortunately, that fix had a problem of it's own which was fixed in wctdm24xxp: Fix FXO failure to detect battery CO disconnects. [2]. This just means there isn't currently a release of the 2.6 branch that contains all the recommended fixes. If you were on the 2.6 branch, then I advise installing the current tip of the 2.6.y branch like: $ git clone git://git.asterisk.org/dahdi/linux dahdi-linux -b 2.6.y $ cd dahdi-linux $ make install [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=41639330a59 [2] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=160edc8c9db If you don't have git installed on the machine you would like to install this on, you can use the 'snapshot' link when looking at the shortlog of the 2.6.y branch at git.asterisk.org [3] which will allow you to download a tar.gz file. [3] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/heads/2.6.y -- _ -- 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] RED on DAHDI channel
I got the following warning during the build. Is it anything to worry about? WARNING: could not find /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o Mitch On 05/28/2013 12:37 PM, Mitch Claborn wrote: I am running 2.6.1. I'll give the 2.6.y a try. Mitch On 05/28/2013 10:53 AM, Shaun Ruffell wrote: On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? Hi Mitch, What version of DAHDI are you using? Unfortunately I did insert a bug on 2.6.0 where it was possible for a channel on an AEX410 to get stuck in RED alarm depending on the timing from the central office. If you're not using 2.6.0+ you can ignore the remainder of this email. The bug was fixed in 2.6.2 in wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. [1] but unfortunately, that fix had a problem of it's own which was fixed in wctdm24xxp: Fix FXO failure to detect battery CO disconnects. [2]. This just means there isn't currently a release of the 2.6 branch that contains all the recommended fixes. If you were on the 2.6 branch, then I advise installing the current tip of the 2.6.y branch like: $ git clone git://git.asterisk.org/dahdi/linux dahdi-linux -b 2.6.y $ cd dahdi-linux $ make install [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=41639330a59 [2] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=160edc8c9db If you don't have git installed on the machine you would like to install this on, you can use the 'snapshot' link when looking at the shortlog of the 2.6.y branch at git.asterisk.org [3] which will allow you to download a tar.gz file. [3] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/heads/2.6.y -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RED on DAHDI channel
The 2.6.y version installed without issue. A few test calls went OK. Will leave it in and see how things go. The problem has been sporadic, so won't know for a while if the issue is solved. Mitch On 05/28/2013 01:37 PM, Shaun Ruffell wrote: On Tue, May 28, 2013 at 12:44:47PM -0500, Mitch Claborn wrote: I got the following warning during the build. Is it anything to worry about? WARNING: could not find /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o No, that's normal. It's a side effect that the compiler doesn't know all the options that were used to produce the precompile VPMADT032 loader. -- _ -- 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] RED on DAHDI channel
Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? -- Mitch -- _ -- 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] Call stuck in queue
Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get stuck in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log file after lunch and see what I can see, but this is all new territory for me. *queue show* sales has 1 calls (max unlimited) in 'leastrecent' strategy (27s holdtime, 181s talktime), W:0, C:3963, A:28, SL:0.0% within 0s Members: SIP/Eileen (ringinuse disabled) (dynamic) (Not in use) has taken 2 calls (last was 239 secs ago) SIP/KWakmn (ringinuse disabled) (dynamic) (Not in use) has taken 1 calls (last was 502 secs ago) SIP/Britne (ringinuse disabled) (dynamic) (Not in use) has taken 1 calls (last was 365 secs ago) SIP/Kim (ringinuse disabled) (dynamic) (Not in use) has taken 2 calls (last was 84 secs ago) SIP/Charlie (ringinuse disabled) (dynamic) (In use) has taken 4 calls (last was 1438 secs ago) SIP/Carlene (ringinuse disabled) (dynamic) (In use) has taken no calls yet SIP/Erin (ringinuse disabled) (dynamic) (Not in use) has taken 13 calls (last was 1079 secs ago) SIP/Phyllis (ringinuse disabled) (dynamic) (In use) has taken 6 calls (last was 1052 secs ago) SIP/JackieA (ringinuse disabled) (dynamic) (In use) has taken 12 calls (last was 552 secs ago) SIP/Peggy (ringinuse disabled) (dynamic) (In use) has taken 1 calls (last was 822 secs ago) Callers: *1. DAHDI/i1/9705541916-1507 (wait: 4:32, prio: 0)* *core show channels concise * SIP/KWakmn-181a!LocalSets!sales!1!Ringing!AppQueue!(Outgoing Line)!214!!!3!1!(None)!sip1-1367428777.13318 DAHDI/i1/7812693000-1508!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!7812693000!!!3!277!SIP/Peggy-180c!sip1-1367428501.13296 SIP/Erin-1819!LocalSets!sales!1!Ringing!AppQueue!(Outgoing Line)!233!!!3!8!(None)!sip1-1367428769.13317 DAHDI/49-1!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!8438080641!!!3!647!SIP/Charlie-17fa!sip1-1367428130.13268 SIP/Charlie-17fa!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!236!!!3!616!DAHDI/49-1!sip1-1367428162.13271 SIP/JackieA-1814!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!277!!!3!124!DAHDI/51-1!sip1-1367428653.13310 *DAHDI/i1/9705541916-1507!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!9705541916!!!3!319!(None)!sip1-1367428459.13293* DAHDI/50-1!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!6039445485!!!3!372!SIP/Phyllis-1807!sip1-1367428406.13288 DAHDI/i1/5153523240-150b!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!5153523240!!!3!46!(None)!sip1-1367428731.13314 DAHDI/i1/5036635064-1503!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!5036635064!!!3!517!SIP/Carlene-1809!sip1-1367428260.13279 DAHDI/i1/7609539399-150c!queues!sales!7!Up!Read!MMSURVEY,custom/survey-ask,1,,,5!7609539399!!!3!27!(None)!sip1-1367428750.13316 DAHDI/51-1!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!3038412549!!!3!158!SIP/JackieA-1814!sip1-1367428619.13307 SIP/Peggy-180c!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!229!!!3!239!DAHDI/i1/7812693000-1508!sip1-1367428539.13299 SIP/Phyllis-1807!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!213!!!3!338!DAHDI/50-1!sip1-1367428440.13290 SIP/Carlene-1809!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!232!!!3!302!DAHDI/i1/5036635064-1503!sip1-1367428475.13294 *queues.conf* -- [general] autofill=yes ; distribute all waiting callers to available members shared_lastcall=yes ; respect the wrapup time for members logged into more than one queue [StandardQueue](!) ; template to provide common features musicclass=default ; play [default] music strategy=leastrecent ; changed from rrmemory to leastrecent MLC 3/29/2013 joinempty=no ; do not join the queue when no members available leavewhenempty=yes ; leave the queue when no members available ringinuse=no ; don't ring members when already InUse (prevents multiple calls to an agent) announce-frequency = 30 min-announce-frequency = 15 announce-holdtime = yes|no|once announce-position = limit announce-position-limit = 5 announce-round-seconds = 10 setinterfacevar = yes ; set some variables setqueueentryvar = yes ; some more variables setqueuevar = yes ; some more variables wrapuptime=2 ; wait 2 seconds before next call timeout = 16 ; try each operator for 12 seconds before moving to the next one autopause=no ; don't pause a member when they fail to answer a call [sales](StandardQueue) ; create the sales queue using the parameters in the StandardQueue template -- Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live
Re: [asterisk-users] Call stuck in queue
Mystery mostly solved. One of the operators failed to log out of the queue when she went to lunch. Because of the leastrecent strategy, when that operator was the least recent, it kept trying her until she was no longer least recent and then tried a different operator. About to experiment with autopause and autopausedelay to see if that will help. Mitch On 05/01/2013 01:11 PM, Mitch Claborn wrote: Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get stuck in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log file after lunch and see what I can see, but this is all new territory for me. *queue show* sales has 1 calls (max unlimited) in 'leastrecent' strategy (27s holdtime, 181s talktime), W:0, C:3963, A:28, SL:0.0% within 0s Members: SIP/Eileen (ringinuse disabled) (dynamic) (Not in use) has taken 2 calls (last was 239 secs ago) SIP/KWakmn (ringinuse disabled) (dynamic) (Not in use) has taken 1 calls (last was 502 secs ago) SIP/Britne (ringinuse disabled) (dynamic) (Not in use) has taken 1 calls (last was 365 secs ago) SIP/Kim (ringinuse disabled) (dynamic) (Not in use) has taken 2 calls (last was 84 secs ago) SIP/Charlie (ringinuse disabled) (dynamic) (In use) has taken 4 calls (last was 1438 secs ago) SIP/Carlene (ringinuse disabled) (dynamic) (In use) has taken no calls yet SIP/Erin (ringinuse disabled) (dynamic) (Not in use) has taken 13 calls (last was 1079 secs ago) SIP/Phyllis (ringinuse disabled) (dynamic) (In use) has taken 6 calls (last was 1052 secs ago) SIP/JackieA (ringinuse disabled) (dynamic) (In use) has taken 12 calls (last was 552 secs ago) SIP/Peggy (ringinuse disabled) (dynamic) (In use) has taken 1 calls (last was 822 secs ago) Callers: *1. DAHDI/i1/9705541916-1507 (wait: 4:32, prio: 0)* *core show channels concise * SIP/KWakmn-181a!LocalSets!sales!1!Ringing!AppQueue!(Outgoing Line)!214!!!3!1!(None)!sip1-1367428777.13318 DAHDI/i1/7812693000-1508!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!7812693000!!!3!277!SIP/Peggy-180c!sip1-1367428501.13296 SIP/Erin-1819!LocalSets!sales!1!Ringing!AppQueue!(Outgoing Line)!233!!!3!8!(None)!sip1-1367428769.13317 DAHDI/49-1!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!8438080641!!!3!647!SIP/Charlie-17fa!sip1-1367428130.13268 SIP/Charlie-17fa!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!236!!!3!616!DAHDI/49-1!sip1-1367428162.13271 SIP/JackieA-1814!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!277!!!3!124!DAHDI/51-1!sip1-1367428653.13310 *DAHDI/i1/9705541916-1507!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!9705541916!!!3!319!(None)!sip1-1367428459.13293* DAHDI/50-1!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!6039445485!!!3!372!SIP/Phyllis-1807!sip1-1367428406.13288 DAHDI/i1/5153523240-150b!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!5153523240!!!3!46!(None)!sip1-1367428731.13314 DAHDI/i1/5036635064-1503!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!5036635064!!!3!517!SIP/Carlene-1809!sip1-1367428260.13279 DAHDI/i1/7609539399-150c!queues!sales!7!Up!Read!MMSURVEY,custom/survey-ask,1,,,5!7609539399!!!3!27!(None)!sip1-1367428750.13316 DAHDI/51-1!queues!sales!21!Up!Queue!sales,tc,,sub-QueueConnected!3038412549!!!3!158!SIP/JackieA-1814!sip1-1367428619.13307 SIP/Peggy-180c!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!229!!!3!239!DAHDI/i1/7812693000-1508!sip1-1367428539.13299 SIP/Phyllis-1807!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!213!!!3!338!DAHDI/50-1!sip1-1367428440.13290 SIP/Carlene-1809!LocalSets!sales!1!Up!AppQueue!(Outgoing Line)!232!!!3!302!DAHDI/i1/5036635064-1503!sip1-1367428475.13294 *queues.conf* -- [general] autofill=yes ; distribute all waiting callers to available members shared_lastcall=yes ; respect the wrapup time for members logged into more than one queue [StandardQueue](!) ; template to provide common features musicclass=default ; play [default] music strategy=leastrecent ; changed from rrmemory to leastrecent MLC 3/29/2013 joinempty=no ; do not join the queue when no members available leavewhenempty=yes ; leave the queue when no members available ringinuse=no ; don't ring members when already InUse (prevents multiple calls to an agent) announce-frequency = 30 min-announce-frequency = 15 announce-holdtime = yes|no|once announce-position = limit announce-position-limit = 5 announce-round-seconds = 10 setinterfacevar = yes ; set some variables setqueueentryvar = yes ; some more variables setqueuevar = yes ; some more variables wrapuptime=2 ; wait 2 seconds before next call timeout = 16 ; try each operator for 12 seconds before moving
Re: [asterisk-users] Asterisk 11 -CDR values changed in hangup handler not saved ?
I have seen that behavior also. Mitch On 03/28/2013 06:56 PM, Olivier wrote: Hello, I'm using Hanhup Handlers in a testing asterisk 11 system. Within one such handler, I'm setting CDR values. To me, it seems those changed CDR values are not saved in CDR back-end. Can you confirm ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] sip set debug on output to file only (not to console)
I recently faced the same issue. I didn't find a way in Asterisk to do what I wanted. A good workaround is to use wireshark in batch mode (tshark) to trace traffic to the IP address you are interested in. You should be able to filter it to capture only SIP traffic. Mitch On 03/29/2013 08:02 AM, Marie Fischer wrote: Hello everybody, I am trying to find an intermittent SIP error with one provider and thought the best first step would be to have sip set debug on for some days and check the logs. Everything gets logged nicely, but the SIP log clutters up the console quite badly. Is it possible to have SIP debug log go only to the log file and not to the console? My logger.conf: console = notice,warning,error messages = notice,warning,error full = notice,warning,error,debug,verbose,dtmf,fax On the console, I entered: core set verbose 3 core set debug 0 sip set debug on Thanks, -- _ -- 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 11 -CDR values changed in hangup handler not saved ?
My personal opinion is that it is a design flaw. It is probably working as designed, but I think the design should be different. I did not find any workaround. Mitch On 03/29/2013 11:14 AM, Olivier wrote: How would you qualify it ? A feature ? A bug ? Could you find a work around ? 2013/3/29 Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net I have seen that behavior also. Mitch On 03/28/2013 06:56 PM, Olivier wrote: Hello, I'm using Hanhup Handlers in a testing asterisk 11 system. Within one such handler, I'm setting CDR values. To me, it seems those changed CDR values are not saved in CDR back-end. Can you confirm ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/__mailman/listinfo/asterisk-__users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/__mailman/listinfo/asterisk-__users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing call problem
I've installed 7 Digium D40's over the last 24 hours. They work flawlessly - no dropped calls, no 1-way audio, sound quality is noticeably better. If these work out through Monday (our busiest day) then we'll order a dozen more for the rest of the agents. The one downside to this approach is that the agent has to have one headset for the phone and another for their computer (which they need occasionally). I get to go home on Saturday! The Digium phone deployment is simple enough to manage remotely. Mitch On 03/22/2013 01:13 PM, Matthew J. Roth wrote: Mitch Claborn wrote: Interestingly, using Bria we sometimes see similar, though not exactly the same, symptoms. That would make me wonder about the TCP stack on the client machine, or similar. With a softphone, you're dependent on the entire software stack up to the softphone and at the mercy of every other process. They're often a cheaper solution, but the trade-off comes in the form of reliability and stability. We are close to ditching the soft phones entirely for this call center and going to the Digium D40. I put one of those in service this morning and the calls are noticeably clearer and there have been no reported problems. The hardphone eliminates a lot of variables, so it's a very good idea to at least use them in your test environment. Using them in production may be more expensive at first, but if they're easier to manage then they may be more economical in the long run. Good luck and I hope you get to go home this weekend. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing call problem
I did open a ticket with SFL support and sent them the packet trace. Interestingly, using Bria we sometimes see similar, though not exactly the same, symptoms. That would make me wonder about the TCP stack on the client machine, or similar. Bria on Ubuntu is not terribly stable. Bria on the Mac works very well, but that's a pretty expensive solution. We are close to ditching the soft phones entirely for this call center and going to the Digium D40. I put one of those in service this morning and the calls are noticeably clearer and there have been no reported problems. Mitch On 03/21/2013 09:48 AM, Matthew J. Roth wrote: Mitch Claborn wrote: Thank you for that most excellent post. I had guessed at most of the SDP fields and meaning. No problem. I actually like looking at SIP traces for some reason. I have wireshark traces from the client and the RTP packets are not in the trace, which I think means that the client software is simply not producing them. I have opened a ticket with SFL phone support and will post here if I find anything. That's a reasonable conclusion. Just make sure that you get some traces of good calls to verify that your tests are valid. I did test the muted microphone theory. SFLphone continues to send RTP packets even when the mic is muted, so that doesn't seem to be the cause. It's always a good idea to rule out PEBKAC before spending a lot of time diagnosing a problem. I've also compared the call initiation SIP and SDP packets between a call that fails and one that works correctly. I can discern no difference other than things like port numbers and call IDs. Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe that will make a difference. It really seems like it may be a problem with the softphone. I'm sure the developers of SFLphone will appreciate your feedback, because not sending RTP is a pretty serious bug. I'll keep an eye on this thread and help out if I can. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing call problem
Version (v): 0 Owner/Creator, Session Id (o): asset071 3572788447 1 IN IP4 172.16.0.71 Session Name (s): sflphone Connection Information (c): IN IP4 172.16.0.71 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 45208 RTP/AVP 0 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): sendrecv Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): rtcp:45209 IN IP4 172.16.0.71 -- No. TimeSourceDestination Protocol Length Info 6 12:14:18.056116 172.16.0.245 172.16.0.71 SIP 463Request: ACK sip:KWakmn@172.16.0.71:5060 Frame 6: 463 bytes on wire (3704 bits), 463 bytes captured (3704 bits) Ethernet II, Src: 90:b1:1c:0d:c4:35 (90:b1:1c:0d:c4:35), Dst: Dell_e7:fc:b0 (00:25:64:e7:fc:b0) Internet Protocol Version 4, Src: 172.16.0.245 (172.16.0.245), Dst: 172.16.0.71 (172.16.0.71) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: ACK sip:KWakmn@172.16.0.71:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK5a4eafc5 Max-Forwards: 70 From: sip:4062345243@172.16.0.245;tag=as5a63ac9a To: sip:KWakmn@172.16.0.71:5060;tag=923b9add-5ef2-4f6f-a3f5-4627109e079c Contact: sip:4062345243@172.16.0.245:5060 Call-ID: 50b7a1e27bbb9f6043dfccff16d7be88@172.16.0.245:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 11.1.0 Content-Length: 0 -- Mitch On 03/19/2013 07:18 PM, Mitch Claborn wrote: Good point. I changed to 1 - 4. Mitch On 03/19/2013 06:17 PM, Asghar Mohammad wrote: i had this problem with a gateway witch was configured from 1000 to 3000 and some time he was using ports above 2000 and result was one way voice rtp port range is where asterisk expect audio, you should not use ports below 1 because they are in use of other services like 5060 for sip. On Tue, Mar 19, 2013 at 11:57 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: This was the client sending from port 39409 to server port 13429, which is in the range. From what I read, the rtpstart and rtpend define the range that is available for use on the server, so I'm not sure this will apply. But, I've set my range to 5000 - 4. I'll find out tomorrow if it makes any difference. Where is a good place to find documentation on the various fields in the INVITE SIP message and the response? I'd like to dig into them and try to understand them more completely. Mitch On 03/19/2013 05:02 PM, Asghar Mohammad wrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy from asterisk 11 rtp.conf rtpstart=1 rtpend=2 have you changed port range? if no then your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port. try to change rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad asghar...@gmail.com mailto:asghar...@gmail.com mailto:asghar...@gmail.com mailto:asghar...@gmail.com wrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy from asterisk 11 rtp.conf rtpstart=1 rtpend=2 have you changed port range? if no then your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port. try to change rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net wrote: We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3. There is no NAT involved in the network at all (it is disabled in sip.conf). Here are the SIP messages capture via wireshark on the client during one problem call. Following these SIP messages, the wireshark trace shows only RTP packets from server (172.16.0.245) to client (172.16.0.71) except for an occasional RTCP packet from client to server (sample below
Re: [asterisk-users] Diagnosing call problem
There is no firewall on the client. I've compared the SIP messages between a successful call and a failed call, and I can see no difference except for things like port numbers and call IDs. It only fails occasionally, not on every call. Mitch On 03/20/2013 01:16 PM, Asghar Mohammad wrote: On Wed, Mar 20, 2013 at 7:11 PM, Asghar Mohammad asghar...@gmail.com mailto:asghar...@gmail.com wrote: hi, problem seem to client end i am going to install SFLPhone i will let you know when finish, have you check firewall on clients pc? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing call problem
Thank you for that most excellent post. I had guessed at most of the SDP fields and meaning. I have wireshark traces from the client and the RTP packets are not in the trace, which I think means that the client software is simply not producing them. I have opened a ticket with SFL phone support and will post here if I find anything. I did test the muted microphone theory. SFLphone continues to send RTP packets even when the mic is muted, so that doesn't seem to be the cause. I've also compared the call initiation SIP and SDP packets between a call that fails and one that works correctly. I can discern no difference other than things like port numbers and call IDs. Tomorrow I'll be trying one of my agents on Bria instead of SFL - maybe that will make a difference. Mitch On 03/20/2013 02:09 PM, Matthew J. Roth wrote: Mitch Claborn wrote: Where is a good place to find documentation on the various fields in the INVITE SIP message and the response? I'd like to dig into them and try to understand them more completely. For the SIP headers: http://en.wikipedia.org/wiki/Session_Initiation_Protocol http://www.ietf.org/rfc/rfc3261.txt For the SDP content: http://en.wikipedia.org/wiki/Session_Description_Protocol http://www.ietf.org/rfc/rfc4566.txt Don't forget that SIP is a request-response protocol. The server sends an INVITE with SDP describing the media session on its end (RTP IP and port, codec, etc.) but that only gives you half of the picture. The client returns an OK with SDP describing its side of the media session. You have to look at both to determine if the call was negotiated properly. To start, I'm going to strip down one of the SIP traces you sent so it's not overwhelming: INVITE from Asterisk server (172.16.0.245) to client (172.16.0.71) c=IN IP4 172.16.0.245 m=audio 13428 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv This says that the Asterisk server's RTP for the call will be at 172.16.0.245 (from the c= line) port 13428 (from the m= line), the allowed codecs are u-law (0 PCMU), a-law (8 PCMA), and DTMF (101 telephone-event) (from the m= and a= lines), and Asterisk will both send and receive packets. Note that this is the port (13428) that must be within the range defined in rtp.conf. The port returned in the client's OK is specific to the client and Asterisk has no control over it. Speaking of the client's OK: OK from client (172.16.0.71) to Asterisk server (172.16.0.245) c=IN IP4 172.16.0.71 m=audio 39408 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 This says that the client's RTP for the call will be at 172.16.0.71 (from the c= line) port 39408 (from the m= line), the allowed codec is u-law (0 PCMU) (from the m= and a= lines), and the client will both send and receive packets. There is also a stray a= line describing DTMF, but its payload type (101) isn't listed on the m= line. I may be wrong, but that seems broken to me. I don't think it would cause the audio issues you're describing, but it's something you could ask SFLphone support about. So the IPs and ports are agreed on (Asterisk = 172.16.0.245:13428, client = 172.16.0.71:39408), both endpoints share an allowed codec (u-law), and they're both ready to send and receive packets. The good news is that the call should work. The bad news is it doesn't. The RTCP information you posted bears this out: Fraction lost: 254 / 256 Cumulative number of packets lost: 37134 Extended highest sequence number received: 37331 Over 99% of the packets are lost, so the call is setup fine but something is getting in the way of the RTP. Your first post said: Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording. This means that the agent's RTP never makes it to the Asterisk process. I doubt it's even making it to the server, but you could prove it by running: # tcpdump -s 0 -A host 172.16.0.71 and portrange 1-65535 at the Linux command line during a bad call. If you only see packets going to the client that takes your Asterisk configuration out of the equation. Then you have to start tracing it back to the client. First rule out the firewall on the Asterisk server, then the cable to the switch, then the switch, then the cable to the client, then the client's firewall, then the softphone on the client. Something on that path has to be stopping (or not producing) the agent's RTP. Don't forget the simple stuff either. It could be something like the agent putting their microphone on mute. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer
Re: [asterisk-users] Diagnosing call problem
I don't believe the headsets are at fault. An agent will have a number of calls that work just fine, then with no apparent change by the agent, a few calls in a row will not work. In some cases, the problem seems to correct itself. In other cases, restarting the agent's computer seems to fix the problem. Mitch On 03/18/2013 11:51 PM, Satish Barot wrote: On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording. Looking for ideas on how to begin to diagnose this or clues about what might be wrong. Is there a console command that will show details of a specific call in progress that might have some clues? -- Mitch -- _ -- 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 http://lists.digium.com/mailman/listinfo/asterisk-users Silly guess, If there is no then NAT did you check that your headphones work properly every time you start the softphone? This has happened to me in past. --Satish Barot Ahmedabad, India. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing call problem
Thanks for the suggestions. 1) directmedia was taking the default of yes. I set to no. Will watch and see. 2) NAT is turned off (nat=no). I've never done any RTP debugging. Is that rtp set debug on ip 1.2.3.4? How would I interpret the output? 3) mixmonitor recordings are stored on a local disk (RAID array, very fast) 4) This would have to be a last resort option, as there is a business requirement to record the agent calls Mitch On 03/19/2013 12:01 AM, Bharat Lalcheta wrote: 1) Check directmedia option in sip. If enabled set it to no 2) Check NAT option and RTP debug in live scenario for any particular agent 3) if not solved yet, Where are your storing your mixmonitor recording? On any storage ? If yes, try to record on local harddisk. 4) Remove mixmonitor and test again Hope you find can find problem 99% in above scenario. Regards, Bharat Lalcheta On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot satish4aster...@gmail.com mailto:satish4aster...@gmail.com wrote: On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording. Looking for ideas on how to begin to diagnose this or clues about what might be wrong. Is there a console command that will show details of a specific call in progress that might have some clues? -- Mitch -- _ -- 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 http://lists.digium.com/mailman/listinfo/asterisk-users Silly guess, If there is no then NAT did you check that your headphones work properly every time you start the softphone? This has happened to me in past. --Satish Barot Ahmedabad, India. -- _ -- 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 -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Diagnosing call problem
Identifier: 0x841ef2ea (2216620778) SDES items Type: CNAME (user and domain) (1) Length: 17 Text: kristin@localhost Type: END (0) [RTCP frame length check: OK - 60 bytes] Mitch On 03/19/2013 12:02 PM, Asghar Mohammad wrote: witch softphone you are using? on client pc installed some kind of virtualpc like vmware or virtualbox? client pc have more then one network interfaces? you can capture sip invites from soft phone by enabling debug on client ip sip set debug ip ip of softphon upload sip trace then somebody can halp you, should provide more information's. On Tue, Mar 19, 2013 at 5:39 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: rtp debug on the calls that do not work correctly shows packets from server to client only, none from client to server. I do have nat=no directmedia=no in sip.conf. Are there other settings that might apply? This last instance that I looked at, the problem persisted even after restarting the client softphone program. It was fixed after rebooting the client computer. Any ideas on a next step for debugging? I was thinking I would start a wireshark trace to see if the rtp packets are actually leaving the client computer. Mitch On 03/19/2013 08:28 AM, Bharat Lalcheta wrote: rtp set debug ip 1.2.3.4 where 1.2.3.4 is ip of your particular agent. Say your x agent is not getting voice, rtp debu his ip. You got rtp packet from and to for that ip. If you find rtp packet from your agent to your server ip and rtp packet from your server to agent ip, then no need to check anything in asterisk. Its related to your agent pc problem If you find any single side rtp, then its problem related to nat or direct media etc. if mix monitor is on storage than only you can face problem and thats also very rare. In that case you get voice in break, but it will be from both side not in single side. So, this is not your problem at all. Hope you will get something in rtp debug. R u using any trunk then also check rtp debug between your server and trunk regards, Bharat Lalcheta On Tue, Mar 19, 2013 at 6:39 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net wrote: Thanks for the suggestions. 1) directmedia was taking the default of yes. I set to no. Will watch and see. 2) NAT is turned off (nat=no). I've never done any RTP debugging. Is that rtp set debug on ip 1.2.3.4? How would I interpret the output? 3) mixmonitor recordings are stored on a local disk (RAID array, very fast) 4) This would have to be a last resort option, as there is a business requirement to record the agent calls Mitch On 03/19/2013 12:01 AM, Bharat Lalcheta wrote: 1) Check directmedia option in sip. If enabled set it to no 2) Check NAT option and RTP debug in live scenario for any particular agent 3) if not solved yet, Where are your storing your mixmonitor recording? On any storage ? If yes, try to record on local harddisk. 4) Remove mixmonitor and test again Hope you find can find problem 99% in above scenario. Regards, Bharat Lalcheta On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot satish4aster...@gmail.com mailto:satish4aster...@gmail.com mailto:satish4asterisk@gmail.__com mailto:satish4aster...@gmail.com mailto:satish4asterisk@gmail. mailto:satish4asterisk@gmail.com mailto:satish4asterisk@gmail.__com mailto:satish4aster...@gmail.com wrote: On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net__ wrote: Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can
Re: [asterisk-users] Diagnosing call problem
This was the client sending from port 39409 to server port 13429, which is in the range. From what I read, the rtpstart and rtpend define the range that is available for use on the server, so I'm not sure this will apply. But, I've set my range to 5000 - 4. I'll find out tomorrow if it makes any difference. Where is a good place to find documentation on the various fields in the INVITE SIP message and the response? I'd like to dig into them and try to understand them more completely. Mitch On 03/19/2013 05:02 PM, Asghar Mohammad wrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy from asterisk 11 rtp.conf rtpstart=1 rtpend=2 have you changed port range? if no then your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port. try to change rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad asghar...@gmail.com mailto:asghar...@gmail.com wrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy from asterisk 11 rtp.conf rtpstart=1 rtpend=2 have you changed port range? if no then your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port. try to change rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3. There is no NAT involved in the network at all (it is disabled in sip.conf). Here are the SIP messages capture via wireshark on the client during one problem call. Following these SIP messages, the wireshark trace shows only RTP packets from server (172.16.0.245) to client (172.16.0.71) except for an occasional RTCP packet from client to server (sample below). Any help is appreciated. The uses are really beating me up to get this fixed. INVITE sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.71:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.245:5060;branch=__z9hG4bK19e2246d Max-Forwards: 70 From: sip:2392230612@172.16.0.245 mailto:sip%3A2392230612@172.16.0.245;__tag=as4b489afc To: sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.71:5060 Contact: sip:2392230612@172.16.0.245:__5060 http://sip:2392230612@172.16.0.245:5060 Call-ID: 52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060 http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.1.0 Date: Tue, 19 Mar 2013 20:47:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-mm-call: http://www.mcmurrayhatchery.__com http://www.mcmurrayhatchery.com Content-Type: application/sdp Content-Length: 257 v=0 o=root 682517197 682517197 IN IP4 172.16.0.245 s=Asterisk PBX 11.1.0 c=IN IP4 172.16.0.245 t=0 0 m=audio 13428 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --__- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.16.0.245:5060;received=__172.16.0.245;branch=__z9hG4bK19e2246d Call-ID: 52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060 http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060 From: sip:2392230612@172.16.0.245 mailto:sip%3A2392230612@172.16.0.245;__tag=as4b489afc To: sip:KWakmn@172.16.0.71 mailto:sip%3AKWakmn@172.16.0.71;tag=__7543f39a-7ca0-434b-8281-__e6dc2adc4aa3 CSeq: 102 INVITE Contact: sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.71:5060 Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE, INVITE, ACK, BYE, CANCEL Content-Length: 0 --__--- SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.0.245:5060;received=__172.16.0.245;branch=__z9hG4bK19e2246d Call-ID: 52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060 http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060 From: sip:2392230612@172.16.0.245 mailto:sip%3A2392230612@172.16.0.245;__tag=as4b489afc To: sip:KWakmn@172.16.0.71 mailto:sip%3AKWakmn
Re: [asterisk-users] Diagnosing call problem
The network is all on a single LAN segment - there is no NAT involved anywhere. Agents do not have firewall or active anti-virus. See other posts for a SIP trace. Mitch On 03/19/2013 12:45 PM, Bharat Lalcheta wrote: Firewall can cause problem on client side. Check antivirus or firewall on agent pc Please provide your network setup for getting better idea of problem On Mar 19, 2013 10:10 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: rtp debug on the calls that do not work correctly shows packets from server to client only, none from client to server. I do have nat=no directmedia=no in sip.conf. Are there other settings that might apply? This last instance that I looked at, the problem persisted even after restarting the client softphone program. It was fixed after rebooting the client computer. Any ideas on a next step for debugging? I was thinking I would start a wireshark trace to see if the rtp packets are actually leaving the client computer. Mitch On 03/19/2013 08:28 AM, Bharat Lalcheta wrote: rtp set debug ip 1.2.3.4 where 1.2.3.4 is ip of your particular agent. Say your x agent is not getting voice, rtp debu his ip. You got rtp packet from and to for that ip. If you find rtp packet from your agent to your server ip and rtp packet from your server to agent ip, then no need to check anything in asterisk. Its related to your agent pc problem If you find any single side rtp, then its problem related to nat or direct media etc. if mix monitor is on storage than only you can face problem and thats also very rare. In that case you get voice in break, but it will be from both side not in single side. So, this is not your problem at all. Hope you will get something in rtp debug. R u using any trunk then also check rtp debug between your server and trunk regards, Bharat Lalcheta On Tue, Mar 19, 2013 at 6:39 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net wrote: Thanks for the suggestions. 1) directmedia was taking the default of yes. I set to no. Will watch and see. 2) NAT is turned off (nat=no). I've never done any RTP debugging. Is that rtp set debug on ip 1.2.3.4? How would I interpret the output? 3) mixmonitor recordings are stored on a local disk (RAID array, very fast) 4) This would have to be a last resort option, as there is a business requirement to record the agent calls Mitch On 03/19/2013 12:01 AM, Bharat Lalcheta wrote: 1) Check directmedia option in sip. If enabled set it to no 2) Check NAT option and RTP debug in live scenario for any particular agent 3) if not solved yet, Where are your storing your mixmonitor recording? On any storage ? If yes, try to record on local harddisk. 4) Remove mixmonitor and test again Hope you find can find problem 99% in above scenario. Regards, Bharat Lalcheta On Tue, Mar 19, 2013 at 10:21 AM, Satish Barot satish4aster...@gmail.com mailto:satish4aster...@gmail.com mailto:satish4asterisk@gmail.__com mailto:satish4aster...@gmail.com mailto:satish4asterisk@gmail. mailto:satish4asterisk@gmail.com mailto:satish4asterisk@gmail.__com mailto:satish4aster...@gmail.com wrote: On Tue, Mar 19, 2013 at 12:00 AM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net__ wrote: Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard
Re: [asterisk-users] Diagnosing call problem
Good point. I changed to 1 - 4. Mitch On 03/19/2013 06:17 PM, Asghar Mohammad wrote: i had this problem with a gateway witch was configured from 1000 to 3000 and some time he was using ports above 2000 and result was one way voice rtp port range is where asterisk expect audio, you should not use ports below 1 because they are in use of other services like 5060 for sip. On Tue, Mar 19, 2013 at 11:57 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net wrote: This was the client sending from port 39409 to server port 13429, which is in the range. From what I read, the rtpstart and rtpend define the range that is available for use on the server, so I'm not sure this will apply. But, I've set my range to 5000 - 4. I'll find out tomorrow if it makes any difference. Where is a good place to find documentation on the various fields in the INVITE SIP message and the response? I'd like to dig into them and try to understand them more completely. Mitch On 03/19/2013 05:02 PM, Asghar Mohammad wrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy from asterisk 11 rtp.conf rtpstart=1 rtpend=2 have you changed port range? if no then your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port. try to change rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:54 PM, Asghar Mohammad asghar...@gmail.com mailto:asghar...@gmail.com mailto:asghar...@gmail.com mailto:asghar...@gmail.com wrote: hi, User Datagram Protocol, Src Port: 39409 (39409), Dst Port: 13429 (13429) copy from asterisk 11 rtp.conf rtpstart=1 rtpend=2 have you changed port range? if no then your client sending rtp to a port higher then configured in rtp port range and asterisk ignore that port. try to change rtpend=3 or if there is option in softphone restrict it to use same range as in rtp.conf. let me know if this solve you problem. On Tue, Mar 19, 2013 at 10:22 PM, Mitch Claborn mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net mailto:mitch...@claborn.net wrote: We have Ubuntu 12.04 clients, using either SFLPhone or Bria 3. There is no NAT involved in the network at all (it is disabled in sip.conf). Here are the SIP messages capture via wireshark on the client during one problem call. Following these SIP messages, the wireshark trace shows only RTP packets from server (172.16.0.245) to client (172.16.0.71) except for an occasional RTCP packet from client to server (sample below). Any help is appreciated. The uses are really beating me up to get this fixed. INVITE sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.__71:5060 http://sip:KWakmn@172.16.0.71:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.0.245:5060;branch=z9hG4bK19e2246d Max-Forwards: 70 From: sip:2392230612@172.16.0.245 mailto:sip%3A2392230612@172.16.0.245 mailto:sip%3A2392230612@172.__16.0.245 mailto:sip%253A2392230612@172.16.0.245;__tag=as4b489afc To: sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.71:5060 http://sip:KWakmn@172.16.0.__71:5060 http://sip:KWakmn@172.16.0.71:5060 Contact: sip:2392230612 tel:2392230612@172.16.0.245:__5060 http://sip:2392230612@172.16.__0.245:5060 http://sip:2392230612@172.16.0.245:5060 Call-ID: 52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060 http://52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060 http://__52106f231b41169c7eabd3b43d0fc6__e8@172.16.0.245:5060 http://52106f231b41169c7eabd3b43d0fc6e8@172.16.0.245:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.1.0 Date: Tue, 19 Mar 2013 20:47:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-mm-call: http://www.mcmurrayhatchery.com http://www.mcmurrayhatchery.__com
[asterisk-users] Diagnosing call problem
Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording. Looking for ideas on how to begin to diagnose this or clues about what might be wrong. Is there a console command that will show details of a specific call in progress that might have some clues? -- Mitch -- _ -- 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] Diagnosing call problem
Agents and Asterisk server are in the same network, behind the same firewall, so there is no NAT between agents and the server. The outside calls come in on a T1 fed into the asterisk computer. Mitch On 03/18/2013 01:44 PM, Gertjan Baarda wrote: Is the callcenter sitting behind nat? Sent from my iPhone On 18 mrt. 2013, at 19:31, Mitch Claborn mitch...@claborn.net wrote: Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording. Looking for ideas on how to begin to diagnose this or clues about what might be wrong. Is there a console command that will show details of a specific call in progress that might have some clues? -- Mitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Capture queue agent drop and put caller back in queue
Asterisk 11 Occasionally we will have a partial power outage, or a piece of network equipment will fail, and our queue agents who are on active calls with callers will be disconnected from the caller. What I'd like to do is capture those calls and put them back in the queue (at a high priority) so that we don't lose the caller. I've tried to duplicate the situation in my lab: I have one agent in the queue, a caller dials into the queue, gets connected to the agent then I pull the ethernet cable out of the agent's computer (testing with a softphone) but I don't see anything happen on the asterisk console. core show channels shows the 2 channels still bridged even though the agent is gone. Shouldn't asterisk somehow know when the agent disappears? How can I accomplish my goal? -- Mitch -- _ -- 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] Users list email totals by year .
It would be nice (for me anyway) if the mailing list and forum were combined. Google Groups does this nicely I believe. Mitch On 01/02/2013 08:53 AM, Eric Wieling wrote: I don't use forums as my web browser can't automatically filter the messages for me like my e-mail program can. I stopped participating in the mailing list when it became clear most of the questions were about FreePBX. That seems to have died down a little in recent years. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, January 02, 2013 9:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Users list email totals by year . From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, January 02, 2013 7:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Users list email totals by year . So where has every body else gone? Still here, but mature working systems, still running 1.4.x Doug As the thread said earlier (I think it was Shaun), the response mechanism has moved a good bit into the forums. The users list still is functional for folks who want to contribute but don’t keep a browser window open to monitor the forums. P.S. since the world has now turned twice, Happy New Year to anyone reading. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do *you* test your changes to dialplans ruled by GotoIfTime?
We bypass this problem by storing business hours and holidays in a database table. We use an ODBC call to return whether or not to play the day or night greeting based on the database. We also store the name of a custom greeting file to play. The database is fairly easy to manipulate with test data. Mitch On 12/27/2012 01:46 PM, Ernie Dunbar wrote: This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan was broken by me misspelling the correct dialplan to go to. Then our Boxing Day dialplan was broken when I copied and pasted the correct dialplan from one similar extension number to the other, like this: ; Christmas ; exten = 821192,n,GotoIfTime(9:30-14:00,*,25,dec?ivr-lightspeed-tech-early,s,1) exten = 821192,n,GotoIfTime(8:00-17:00,*,24,dec?ivr-lightspeed-day,s,1) exten = 821192,n,GotoIfTime(*,*,25,dec?ivr-lightspeed-after-hours,s,1) exten = 821190,n,GotoIfTime(9:00-18:00,*,26,dec?ivr-lightspeed-day,s,1) then failed to notice the problem until it was too late. Of course, that only applied on Boxing day and couldn't be noticed earlier, either. I suppose the first problem where I misspelt the dialplan can be solved by testing the dialplan in another extension and modifying the time to now + 2 minutes. But how can I avoid stupid errors in the extension number, when testing by definition requires that I change the extension number to and fro? This appears to boil down to always remember to test it at the time that it becomes relevant. But if I was the kind of person who always remembered to do things at the right time, then there would never be a need for computers to do jobs like this in the first place. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CDR written before hangup extension
asterisk 11.1
Documentation in cdr.conf for endbeforehexten reads:
Normally, CDR's are not closed out until after all extensions are
finished executing. By enabling this option, the CDR will be ended
before executing the h extension and hangup handlers so that CDR
values such as end and billsec may be retrieved inside of of this
extension.
I have explicitly set endbeforehexten=no, yet the CDR records are being
written as soon as the operator hangs up the call (this is in a queue
situation). If I insert Wait() in the dialplan where the comments note
CDR is already written by this point are, I can query the database and
see the CDR record.
My dialplan excerpt is below. Is there a way to force the CDR to not be
written until the end of my dialplan logic? In particular, I want to be
able to store the results of the post call survey in the CDR. I'm using
cdr_adaptive_odbc.
[queues]
; this runs on the caller's channel
exten =sales,1,Verbose(2,${CALLERID(all)} entering the sales queue)
same =n,Answer()
same =n(asksurvey),Read(MMSURVEY,custom/survey-ask,1,,,5)
same =n,MixMonitor(${CHANNEL(uniqueid)}.wav,b)
same =n,Set(CDR(salesqueue_entered)=1)
same =n,Queue(sales,tc,,sub-QueueConnected)
same =n,GotoIf($[${QUEUESTATUS} = CONTINUE]?checksurvey) ; only
go to the survey if we were connected to a call
same =n,Playback(custom/queue-sales-no-operators)
same =n,Hangup()
same =n(checksurvey),GotoIf($[${MMSURVEY} = 1]?survey,s,1)
same =n,Hangup()
exten =h,1,NoOp(When a sales queue call is hung up)
; note CDR is already written by this point
same =n,StopMixMonitor()
same =n,Hangup()
[survey]
exten =s,1,NoOp(Take the survey)
; note CDR is already written by this point
same =n(q1),Read(MMSURVEYQ1,custom/survey-q1,1,,,5)
... more survey here
--
Mitch
--
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[asterisk-users] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way to determine the name of the calling context. -- Mitch -- _ -- 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] MACRO_CONTEXT equivalent for GoSub
Was looking for 1.8 and above. I ended up doing something similar to
what you describe. Not terribly elegant, but it works.
Mitch
On 12/11/2012 04:03 PM, Danny Nicholas wrote:
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable before the gosub so I could track
it. Something like this
Exten = s,n,Set(from=foo)
Exten = s,n,gosub(showfoo,s,1)
Exten = s,n,Set(from=bar)
Exten = s,n,gosub(showfoo,s,1)
[showfoo]
Exten = s,1,verbose(called from ${from})
Exten = s,n,return()
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
Sent: Tuesday, December 11, 2012 3:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] MACRO_CONTEXT equivalent for GoSub
Is there an equivalent of MACRO_CONTEXT for a GoSub? Looking for a way
to determine the name of the calling context.
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[asterisk-users] Asterisk repository for Ubuntu
Is there an Asterisk repository for Ubuntu that has recent versions (e.g. 11)? The standard Ubuntu repository for Ubuntu 12.04 is stick at 1.8. -- Mitch -- _ -- 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] Bypass queue wrapup time
In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- Mitch -- _ -- 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] Bypass queue wrapup time
Asterisk 1.8 Not currently using realtime. Mitch On 10/29/2012 12:19 PM, Danny Nicholas wrote: As I read the queues.conf.sample file I would say no since you would have to set the value to 0 and reload the queue. If you state your asterisk version and whether you're using realtime, someone might offer a solution. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Monday, October 29, 2012 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Bypass queue wrapup time In our sales queue, we have wrapup time set to 15 seconds. When the phones are really busy, the operators would like the ability to bypass that 15 second wait and grab the next call in the queue. Is that possible? How to accomplish? -- _ -- 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 tie orders taken to specific CDR records
Looking at the uniqueid, I get multiple records for some of them. Am I
getting more than one CDR record per call in some cases?
SELECT uniqueid, COUNT(*) FROM asterisk_cdr
GROUP BY uniqueid
HAVING COUNT(*) 2
Mitch
On 10/26/2012 08:34 AM, Bharat Lalcheta wrote:
Every CDR has uniqueid/callid generated and unique between all records.
This callid generated when call arrives on system. And logged in CDR
record as well. You can use it in your dialplan to bind with your order like
exten = s,1,Set(ORDERID=${UNIQUEID})
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[asterisk-users] i extension not triggering
Asterisk 1.8.10.1~dfsg-1ubuntu1
See dial plan code below. When I dial 123 from a phone in this context,
I simply get a busy signal. Why doesn't the i extension get
triggered? Console at verbosity of 10 only shows == Using SIP RTP
CoS mark 5.
[DockPhone]
exten =288,1,NoOp(Dock Phone)
same =n,Dial(${DOCK_RECIPIENTS},30,kt)
include =emergency-services
; the 'i' is not triggering, not sure why
exten =i,1,NoOp(invalid extension from dock phone i)
same =n,Playback(custom/dock-invalid)
same =n,Hangup()
exten =h,1,NoOp(hangup extension from dock phone)
same =n,Hangup()
--
Mitch
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Re: [asterisk-users] i extension not triggering
I set logger.conf to
console =debug,notice,warning,error,verbose
and get the following output:
== Using SIP RTP CoS mark 5
[Oct 25 10:32:53] NOTICE[3501]: chan_sip.c:22622 handle_request_invite:
Call from 'Mitch295' (192.168.5.104:5060) to extension '123' rejected
because extension not found in context 'DockPhone'.
NoOp() is not a typo, and it shows up correctly with dialplan show
DockPhone.
[ Context 'DockPhone' created by 'pbx_config' ]
'444' = 1. NoOp(Dock Phone)
[pbx_config]
2. Dial(${DOCK_RECIPIENTS},30,kt)
[pbx_config]
3. Verbose(2,DIALSTATUS=${DIALSTATUS})
[pbx_config]
4. GotoIf($[${DIALSTATUS} = ANSWER]?good)
[pbx_config]
5. Playback(custom/dock-no-one-available)
[pbx_config]
6. Wait(2)
[pbx_config]
[good] 7. Hangup()
[pbx_config]
'h' =1. NoOp(hangup extension from dock phone)
[pbx_config]
2. Hangup()
[pbx_config]
'i' =1. NoOp(invalid extension from dock phone i)
[pbx_config]
2. Playback(custom/dock-invalid)
[pbx_config]
3. Hangup()
[pbx_config]
Include ='emergency-services'
[pbx_config]
Mitch
On 10/25/2012 10:19 AM, Danny Nicholas wrote:
It would be good to see OP's output, but noop() is essentially the same as
Verbose(), whatever goes in the () is just a comment/message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, October 25, 2012 10:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] i extension not triggering
exten =i,1,NoOp(invalid extension from dock phone i)
Was this a typo?
I believe it should be:
exten = i,1,NoOP()
What does your console output look like?
Doug
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Re: [asterisk-users] i extension not triggering
That does sound quite suspicious. Mitch It looks like you are seeing this issue that was fixed earlier this month: https://issues.asterisk.org/jira/browse/ASTERISK-20455 Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] i extension not triggering
A little more background will help. This is a phone that will be
outside on our receiving dock. When a driver lifts the handset, the
ObiTalk 110 dials 444 automatically. That all works fine and it rings
the phones that it should.
What I'm trying to do with the i extension is give a friendly message
to the driver if he tries to dial something on his own.
ZXX will catch 123 but won't catch others, like a 7 digit local number
for example.
I could put in patterns for most common patterns, but I really just want
i to work, to catch everything but 444 and the emergency numbers.
Mitch
On 10/25/2012 10:42 AM, Danny Nicholas wrote:
Based on the output below, DockPhone is expecting to be reached with a
dialstring of 444. If you change 444 to ZXX, the problem should go away.
I set logger.conf to
console =debug,notice,warning,error,verbose
and get the following output:
== Using SIP RTP CoS mark 5
[Oct 25 10:32:53] NOTICE[3501]: chan_sip.c:22622 handle_request_invite:
Call from 'Mitch295' (192.168.5.104:5060) to extension '123' rejected
because extension not found in context 'DockPhone'.
NoOp() is not a typo, and it shows up correctly with dialplan show
DockPhone.
[ Context 'DockPhone' created by 'pbx_config' ]
'444' = 1. NoOp(Dock Phone)
[pbx_config]
2. Dial(${DOCK_RECIPIENTS},30,kt)
[pbx_config]
3. Verbose(2,DIALSTATUS=${DIALSTATUS})
[pbx_config]
4. GotoIf($[${DIALSTATUS} = ANSWER]?good)
[pbx_config]
5. Playback(custom/dock-no-one-available)
[pbx_config]
6. Wait(2)
[pbx_config]
[good] 7. Hangup()
[pbx_config]
'h' =1. NoOp(hangup extension from dock phone)
[pbx_config]
2. Hangup()
[pbx_config]
'i' =1. NoOp(invalid extension from dock phone i)
[pbx_config]
2. Playback(custom/dock-invalid)
[pbx_config]
3. Hangup()
[pbx_config]
Include ='emergency-services'
[pbx_config]
Mitch
On 10/25/2012 10:19 AM, Danny Nicholas wrote:
It would be good to see OP's output, but noop() is essentially the same as
Verbose(), whatever goes in the () is just a comment/message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, October 25, 2012 10:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] i extension not triggering
exten =i,1,NoOp(invalid extension from dock phone i)
Was this a typo?
I believe it should be:
exten = i,1,NoOP()
What does your console output look like?
Doug
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[asterisk-users] How to tie orders taken to specific CDR records
Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the call ends, when the orders have already been placed. (Even if the CDR was written earlier, could I retrieve the generated PK from it in the dialplan somehow?) Is there some combination of fields in the CDR that might uniquely identify a specific call? Open to any and all ideas. -- Mitch -- _ -- 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] i extension not triggering
Thanks Tony, this helps. Mitch On 10/25/2012 11:24 AM, Tony Mountifield wrote: The 'i' extension is not used when entering a context. You can only enter a context (with Dial(), Goto(), etc), at an extension that exists. If it doesn't exist, the context cannot be entered. The 'i' extension is only used when already in a context, and is mainly for catching unmatched extensions dialled within a Background or WaitExten. See http://www.voip-info.org/wiki/view/Asterisk+i+extension for further details. There has also been discussion about this in the mailing list over the years. Cheers Tony -- _ -- 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] i extension not triggering
DOCK_RECIPIENTS is a long list of 5+ SIP phones, so this won't work.
Mitch
On 10/25/2012 11:31 AM, Danny Nicholas wrote:
BOP! You don't need no stinkin I in this case! Just put this in front of
the Dial()
Exten = 444,2,Gotoif(${DOCK_RECIPIENTS} != 444]?i,1)
This catches anything they dial that isn't the magic 444.
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Re: [asterisk-users] How to tie orders taken to specific CDR records
Danny - good idea. That works for the first report that I'm creating. Another idea I had that I may explore: Create another table keytable with an auto increment PK. When I place the call in the queue, insert a row into keytable and retrieve the generated PK. Put that value into the CDR as a user defined field. Not all calls will have it, but all sales queue call should. I can than tie that value back to the actual order records. Mitch On 10/25/2012 11:21 AM, Danny Nicholas wrote: You have the uniqueID, which is a pseudo timestamp. More useful to your described effort, though would be the answer and end of call fields. Your backend system is going to have the timestamp of when the order was placed, so you just need to address the calls that sandwich that timestamp. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn Sent: Thursday, October 25, 2012 11:19 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to tie orders taken to specific CDR records Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the call ends, when the orders have already been placed. (Even if the CDR was written earlier, could I retrieve the generated PK from it in the dialplan somehow?) Is there some combination of fields in the CDR that might uniquely identify a specific call? Open to any and all ideas. -- _ -- 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] Setting CDR fields in connected macro of Queue command
Trying to set some CDR fields in the connected macro of a queue
command. None of the custom fields I set are stored in the database,
but I can set userfield and it does get set. I think that the macro
runs on the agent's channel, not the caller's, and this might contribute
to the problem.
From the sample below userfield (and its alias operatorid) are saved
in the CDR record, but salesqueue_answered is not.
What am I missing?
Asterisk 1.8.10.1~dfsg-1ubuntu1
same =n,Queue(sales,tc,QueueConnected)
[macro-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
same =n,Set(CDR(salesqueue_answered)=1)
same
=n,Set(OPERATORID=${ODBC_OPERATORID_FROM_ADDRESS(${MEMBERINTERFACE})})
; userfield is mapped to operatorid in cdr_adaptive_odbc because
setting operatorid directly doesn't work here
same =n,Set(CDR(userfield)=${OPERATORID})
--
Mitch
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Re: [asterisk-users] question on softhangup
Dave Platt provided the following answer to a similar question of mine last week. I was trying to use SoftHangup() to prempt a DAHDI line for an emergency call. Here is his reply. That may be due to a common characteristic of PSTN lines (at least, it's common here in the U.S.) By design, most U.S. PSTN lines have a very asymmetrical response to a physical hangup: - If the calling party hangs up, the call is terminated immediately. - If the called party hangs up, and the calling party does not, the line remains live for some time (typically around 30 seconds, I believe). If the called party goes off-hook again during this period, they can resume the call. If I recall correctly, things were designed this way so that the called party could say Oh, hang on, I answered this call in the bedroom and the stuff I need is in the living room, hang up the extension phone, go to another room, pick up the other phone and carry on with the call. If that's what you're running into here - if the line you are trying to SoftHangup() was handing an inbound call - then there may be no good solution. As far as I know, there is no way to force an incoming PSTN call to release the line, other than go on-hook, and wait for 30 seconds to pass. Several possible workarounds, roughly in order of increasing complexity and decreasing reliability: (1) Keep one of your PSTN lines reserved for emergency calls only; remove it from your inbound hunt group and place it in a Dahdi line group of its own (or don't group it at all). (2) Keep one of your PSTN lines reserved for *outbound* calls only; you should be able to SoftHangup() an outbound call within a second or two. (3) Figure out a way to check the PSTN lines that are in use at the time of an emergency - if they're all in use, somehow find one which was in use for an outbound call, and select it as the one to SoftHangup() and dial upon. (4) If you must keep all of your PSTN lines in bidirectional use, you may have to *tell* the parties that the line is needed for an emergency call, and ask them to release the line. Do a barge-in on the channel, play an alert sound, play a message saying Emergency call in progress, please hang up this line immediately, play the alert sound again for a few seconds, SoftHangup(), Wait(2), and then try dialing. Mitch On 10/16/2012 08:59 PM, Jerry Geis wrote: How do I use softhangup through the AMI interface? I am using 1.4.43. Will softhangup hangup a DAHDI channel? I have found that Action: Hangup does not hangup a DAHDI channel only sip. Thanks, jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] core show channels verbose output
At the end of the output for core show channels verbose is a line that reads 4 active calls. Does anyone know how that number is formatted if there are more than 999 active calls? Will it have a comma or not? -- Mitch -- _ -- 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] Semi OT: Program transfer button on MiTel 5330
The built in (non-programmable) transfer button on the MiTel 5330 does a blind transfer. Any ideas on how to make it do an attended transfer instead? Instead of DTMF tones, it seems to send a SIP message to do a transfer. I've been unable to find a way to change what it does. -- Mitch -- _ -- 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] Tips for installing and configuring Digum cards
Last night we did a trial run. I am happy to report that both analog and T1 lines worked well with the config files generated by dahdi_genconf. Had a couple of minor issues that I'll ask about in separate posts. Of course when we got on-site, discovered that customer really has 6 analog lines instead of just 4. Hopefully the card I ordered last night will make it here by Saturday. Mitch On 10/11/2012 09:40 AM, Jeff LaCoursiere wrote: Totally typical. I don't think you will have any issues. j -- _ -- 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] SoftHangup for emergency calls
Setting up a group of analog lines to use for outbound emergency calls
(911). My current dial plan and debug output shown below. It appears
that when the SoftHangup() is executed that the line does not really
hang up. In the case shown, I had reduced the group to a single DAHDI
(analog) channel and dialed in to that number from the outside. You can
see in the output that the SoftHangup() was executed, but the call was
not terminated - the outside caller stayed connected to something.
Caller no longer heard the sounds from the menu he was in, but the call
itself seemed to stay connected.
Asterisk 1.8 on Ubuntu
Any ideas?
[emergency-services]
exten =911,1,Goto(dialpsap,1)
exten =9911,1,Goto(dialpsap,1)
exten =999,1,Goto(dialpsap,1)
exten =112,1,Goto(dialpsap,1)
exten =dialpsap,1,Verbose(1,Call initiated to PSAP!)
same =n(dialit),Dial(${LOCAL}/${EMERGENCY},30)
same =n,Verbose(2,DIALSTATUS=${DIALSTATUS})
same =n,GotoIf($[${DIALSTATUS} = ANSWER]?good)
same =n(hu),SoftHangup(${EMERGENCY_CHANNEL},a)
same =n,Wait(5)
same =n,Goto(dialit)
same =n(good),NoOp(call good)
same =n,Hangup()
== Using SIP RTP CoS mark 5
-- Executing [911@LocalSets:1] Goto(SIP/mlcm800-,
dialpsap,1) in new stack
-- Goto (LocalSets,dialpsap,1)
-- Executing [dialpsap@LocalSets:1] Verbose(SIP/mlcm800-,
1,Call initiated to PSAP!) in new stack
Call initiated to PSAP!
-- Executing [dialpsap@LocalSets:2] Dial(SIP/mlcm800-,
DAHDI/g20/19725232703,30) in new stack
[Oct 11 19:30:13] WARNING[3740]: app_dial.c:2218 dial_exec_full: Unable
to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [dialpsap@LocalSets:3] Verbose(SIP/mlcm800-,
2,DIALSTATUS=CONGESTION) in new stack
== DIALSTATUS=CONGESTION
-- Executing [dialpsap@LocalSets:4] GotoIf(SIP/mlcm800-,
0?good) in new stack
-- Executing [dialpsap@LocalSets:5]
SoftHangup(SIP/mlcm800-, DAHDI/49,a) in new stack
[Oct 11 19:30:13] WARNING[3740]: app_softhangup.c:122 softhangup_exec:
Soft hanging DAHDI/49-1 up.
-- Executing [dialpsap@LocalSets:6] Wait(SIP/mlcm800-,
5) in new stack
== Spawn extension (MainMenu, s, 13) exited non-zero on 'DAHDI/49-1'
-- Hanging up on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Executing [dialpsap@LocalSets:7] Goto(SIP/mlcm800-,
dialit) in new stack
-- Goto (LocalSets,dialpsap,2)
-- Executing [dialpsap@LocalSets:2] Dial(SIP/mlcm800-,
DAHDI/g20/19725232703,30) in new stack
-- Called DAHDI/g20/19725232703
-- DAHDI/49-1 answered SIP/mlcm800-
-- Hanging up on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
== Spawn extension (LocalSets, dialpsap, 2) exited non-zero on
'SIP/mlcm800-'
--
Mitch
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[asterisk-users] Recommendation for extension mapping on inbound T1 line
Converting this customer from a MiTel system to asterisk. Discovered
that the inbound calls from the T1 are going to extension 366. (This
was mapped in the MiTel for some arcane purpose.) The dial plan I am
currently using is shown below. When loading the dial plan, I get this
warning:
WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for
an extension is strongly discouraged and can have unexpected behavior.
Please use '_X.' instead at line 331 of extensions.conf
Question: Do I need to worry about this warning?
I'm a little leery of just using 366 in the dialplan, since the company
we are dealing with is a little flaky.
[from-pstn]
exten =s,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
same =n,Goto(MainMenu,s,1)
exten =_.,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
same =n,Goto(MainMenu,s,1)
--
Mitch
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Re: [asterisk-users] Recommendation for extension mapping on inbound T1 line
The s extension did not catch the incoming call. It was only when I added a specific 366 or the _. wildcard that I was able to capture the incoming call. Mitch On 10/12/2012 10:18 AM, A J Stiles wrote: If (and only if) all the extensions you are using in all your contexts are numeric, then _. is fine. (But you don't really need it anyway in your example, since the s extension in your from-pstn context will already catch the incoming call.) -- _ -- 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] Tips for installing and configuring Digum cards
In case the moderator doesn't approve my post with the attachment, below is a quick and dirty transcription of the order form. Customer connecting equipment: CSU/DSU Circuit: DS1 Line coding: B8ZS Framing: ESF Jack type: RJ48X / Smartjack (will that fit into the Digum card?) ISDN protocol: NI2 Primary D-channel assignment: 24 FAS Incoming port selection: Ascending sequential 1-24 Indound real-time ANI delivery? yes, 10 digits Does anyone see any red flags or things to watch out for in these specs? Anyone configured a similar line that is willing to share your config files? I'm traveling for the next several hours, so apologies if I don't respond right away. Mitch On 10/10/2012 10:34 AM, Mitch Claborn wrote: Tomorrow evening I'll be at a customer site installing 2 Digum cards - a 4 port analog and 2 port T1. I'd appreciate any tips, resources and links that you have that might help if we run into trouble. It will, of course, be fairly late at night and relatively high pressure to get it working, so I'd like to collect as much information in advance as I can. I was able to install and test the analog card in my lab, so not too worried about that. I installed the T1 card in the lab and got the system and Asterisk to recognize it, but had no T1 to test with. -- _ -- 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] Tips for installing and configuring Digum cards
Tomorrow evening I'll be at a customer site installing 2 Digum cards - a 4 port analog and 2 port T1. I'd appreciate any tips, resources and links that you have that might help if we run into trouble. It will, of course, be fairly late at night and relatively high pressure to get it working, so I'd like to collect as much information in advance as I can. I was able to install and test the analog card in my lab, so not too worried about that. I installed the T1 card in the lab and got the system and Asterisk to recognize it, but had no T1 to test with. -- Mitch -- _ -- 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] Tips for installing and configuring Digum cards
I am a complete novice at T1's, etc. What else besides framing and coding do I need to ask about? Mitch On 10/10/2012 10:41 AM, Jose P. Espinal wrote: From my own experience, get sure that the Telco actually gives you the *correct* information about the T1 (framing, coding, etc.). Sometimes Telco's technicians tend to sound very secure of information they have not confirmed. -- _ -- 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] Tips for installing and configuring Digum cards
There is actually only a single T1. When we ordered the card, customer thought there were two, but found out later there is only 1. Mitch On 10/10/2012 11:50 AM, Steve Edwards wrote: What is the relationship between the 2 Ts? NFAS? I've pissed away many an hour trying to (remotely) identify which T is which, which channels are D, etc. Telco techs may number them differently than you think. -- _ -- 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] Calling out on a group of DAHDI lines
Excellent. I'll give it a try. (Now if I just didn't have to wait to get on-site where those lines are to try it. Too bad there isn't a DAHDI emulator for SIP lines.) Mitch On 10/09/2012 10:48 AM, Richard Mudgett wrote: There are lots of things documented in chan_dahdi.conf.sample. The following option will assign channels 1-4 to group 1. ; Logical groups can be assigned to allow outgoing roll-over. Groups range ; from 0 to 63, and multiple groups can be specified. By default the ; channel is not a member of any group. ; ; Note that an explicit empty value for 'group' is invalid, and will not ; override a previous non-empty one. The same applies to callgroup and ; pickupgroup as well. ; group=1 channel = 1-4 Then you can dial from that group of channels: same = n,Dial(DAHDI/g1/5551212) /* * data is ---v * Dial(DAHDI/pseudo[/extension[/options]]) * Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]]) * Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]]) * Dial(DAHDI/ispan[/extension[/options]]) * Dial(DAHDI/[ispan-](g|G|r|R)group#(0-63)[c|rcadance#|d][/extension[/options]]) * * i - ISDN span channel restriction. * Used by CC to ensure that the CC recall goes out the same span. * Also to make ISDN channel names dialable when the sequence number * is stripped off. (Used by DTMF attended transfer feature.) * * g - channel group allocation search forward * G - channel group allocation search backward * r - channel group allocation round robin search forward * R - channel group allocation round robin search backward * * c - Wait for DTMF digit to confirm answer * rcadance# - Set distintive ring cadance number * d - Force bearer capability for ISDN/SS7 call to digital. */ (b) For emergency calls, I want to be able to force one of these lines available if all are in use. Will SoftHangup() do that? If so, do I need to Wait() after a SoftHangup() before trying to use it? SoftHangup() should do what you want for this. You need to have a wait so the soft hangup will have a chance to be recognized. I would also suggest that if you use g1 in your normal dial, you should use the highest channel as your emergency line. That channel will be the last used by the group so an emergency call will be least likely to kick off an established call. Another approach is to attempt to dial the emergency call normally. If the first attempt fails, then kick an established call. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Calling out on a group of DAHDI lines
I found that I had to chmod 666 /dev/dahdi/* to allow asterisk to use the simulation channels. The /dev/dahdi directory seems to be recreated when dahdi starts. Here is what I finally came up with that works for me. system.conf dynamic=loc,1:0,4,0 fxsks=1-4 dynamic=loc,1:1,4,0 fxoks=5-8 loadzone= us defaultzone = us chan_dahdi.conf signalling=fxs_ks context=simulation group=0 channel=1 signalling=fxo_ks context=dummy group=63 channel=5 I can now dial out on group 63 and it rings in the simulation context, which I forward to a SIP phone for testing. Is that what you expected to see? Mitch On 10/09/2012 12:40 PM, Shaun Ruffell wrote: Minor correction below: On Tue, Oct 09, 2012 at 12:32:44PM -0500, Shaun Ruffell wrote: On Tue, Oct 09, 2012 at 11:46:04AM -0500, Mitch Claborn wrote: (Now if I just didn't have to wait to get on-site where those lines are to try it. Too bad there isn't a DAHDI emulator for SIP lines.) You can use dynamic DAHDI spans to simulate this on a single box if you would with DAHDI-Linux 2.6.0+. Something like: In /etc/dahdi/system.conf use: dynamic=loc,1:0,4,0 fxsks=49-52 Should make the above line: fxsks=1-4 dynamic=loc,1:1,4,0 fxoks=53-56 loadzone= us defaultzone = us And in /etc/asterisk/chan_dahdi.conf: signalling=fxs_ks context=pstn group=0 channel=1 channel=2 channel=3 channel=4 signalling=fxo_ks context=simulation group=63 channel=51 channel=52 channel=53 channel=54 Now you can start up asterisk and group 0 will be your normal group and you can answer these lines in the simulation context. Dynamic local spans have been around for awhile but I've only used them on a regular basis since 2.6.0+. Cheers, Shaun -- _ -- 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] Calling out on a group of DAHDI lines
Here's what I came up with. Works find with the simulated DAHDI dynamic
local channels. I'll find out later in the week how it works with real
hardware.
[emergency-services]
exten =911,1,Goto(dialpsap,1)
exten =9911,1,Goto(dialpsap,1) ;
exten =999,1,Goto(dialpsap,1)
exten =112,1,Goto(dialpsap,1)
exten =dialpsap,1,Verbose(1,Call initiated to PSAP!)
same =n(dialit),Dial(${LOCAL}/${EMERGENCY},30)
same =n,Verbose(2,DIALSTATUS=${DIALSTATUS})
same =n,GotoIf($[${DIALSTATUS} = ANSWER]?good)
same =n(hu),SoftHangup(${EMERGENCY_CHANNEL},a)
same =n,Wait(2)
same =n,Goto(dialit)
same =n(good),NoOp(call good)
same =n,Hangup()
Mitch
On 10/09/2012 10:48 AM, Richard Mudgett wrote:
Asterisk 1.8
(a) We will have a group of 4 analog lines into a Digium card that
will
be used for local calls. What is the best way to use those lines as
a
pool for outbound calls? Can I use ChanIsAvail(), listing those 4
channels, and then use the first one returned?
There are lots of things documented in chan_dahdi.conf.sample. The
following option will assign channels 1-4 to group 1.
; Logical groups can be assigned to allow outgoing roll-over. Groups range
; from 0 to 63, and multiple groups can be specified. By default the
; channel is not a member of any group.
;
; Note that an explicit empty value for 'group' is invalid, and will not
; override a previous non-empty one. The same applies to callgroup and
; pickupgroup as well.
;
group=1
channel = 1-4
Then you can dial from that group of channels:
same = n,Dial(DAHDI/g1/5551212)
/*
* data is ---v
* Dial(DAHDI/pseudo[/extension[/options]])
* Dial(DAHDI/channel#[c|rcadance#|d][/extension[/options]])
*
Dial(DAHDI/subdir!channel#[c|rcadance#|d][/extension[/options]])
* Dial(DAHDI/ispan[/extension[/options]])
*
Dial(DAHDI/[ispan-](g|G|r|R)group#(0-63)[c|rcadance#|d][/extension[/options]])
*
* i - ISDN span channel restriction.
* Used by CC to ensure that the CC recall goes out the same span.
* Also to make ISDN channel names dialable when the sequence number
* is stripped off. (Used by DTMF attended transfer feature.)
*
* g - channel group allocation search forward
* G - channel group allocation search backward
* r - channel group allocation round robin search forward
* R - channel group allocation round robin search backward
*
* c - Wait for DTMF digit to confirm answer
* rcadance# - Set distintive ring cadance number
* d - Force bearer capability for ISDN/SS7 call to digital.
*/
(b) For emergency calls, I want to be able to force one of these
lines
available if all are in use. Will SoftHangup() do that? If so, do I
need to Wait() after a SoftHangup() before trying to use it?
SoftHangup() should do what you want for this. You need to have a wait
so the soft hangup will have a chance to be recognized.
I would also suggest that if you use g1 in your normal dial, you should use the
highest channel as your emergency line. That channel will be the last used
by the group so an emergency call will be least likely to kick off an
established
call.
Another approach is to attempt to dial the emergency call normally. If the
first
attempt fails, then kick an established call.
Richard
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[asterisk-users] Calling out on a group of DAHDI lines
Asterisk 1.8 (a) We will have a group of 4 analog lines into a Digium card that will be used for local calls. What is the best way to use those lines as a pool for outbound calls? Can I use ChanIsAvail(), listing those 4 channels, and then use the first one returned? (b) For emergency calls, I want to be able to force one of these lines available if all are in use. Will SoftHangup() do that? If so, do I need to Wait() after a SoftHangup() before trying to use it? -- Mitch -- _ -- 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] Call me now outbound calls in a queue
I'll give this a try today and post the results here. Mitch On 10/04/2012 02:30 PM, Ioan Indreias wrote: Hello Mitch, Hoping that the Queue application is not automatically Answering the line (till an agent will do this) my suggestion is to switch between who have to answer in order to progress to the second call leg. This means that the Queue will be called through a Local Channel and the call to your customer will be made through a Dial application. Below is something to start with - in case it will work you could modify to your needs. [demo] exten = s,1,NoOp(Queue without answer) exten = s,2,Queue(sales) Action: Originate Channel: Local/s@demo/n Application: Dial Data: SIP/voipms/customer_number HTH, Ioan Indreias Modulo Consulting // www.modulo.ro http://www.modulo.ro -- _ -- 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] Call me now outbound calls in a queue
This is mostly working. See below. My only problem is being able to
set the caller ID on the outbound call to the customer. I've tried both
a queue connected macro and gosub (see below), and those both execute,
but the caller ID is not showing up correctly for the customer. I
assume this is because the caller ID is being set on the agent's channel
not the customers.
Any ideas on that?
Action: Originate
Channel: Local/s@callmenow/n
Application: Dial
Data: SIP/voipms/customer_number
Async: true
Callerid: Call Me Now 777
Timeout: 99
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same
=n,Queue(sales,tc,CallMeNowQueueConnected,CallMeNowQueueConnectedGosub)
[CallMeNowQueueConnectedGosub]
exten =s,1,NoOp(CallMeNowQueueConnectedGosub)
same =n,Set(CALLERID(num)=${OUTBOUND_CALLERID_NUM})
same =n,Set(CALLERID(name)=${OUTBOUND_CALLERID_NAME})
same =n,Verbose(2,end of gosub)
same =n,Return()
[macro-CallMeNowQueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp(CallMeNowQueueConnected)
same =n,Set(CALLERID(num)=${OUTBOUND_CALLERID_NUM})
same =n,Set(CALLERID(name)=${OUTBOUND_CALLERID_NAME})
same =n,Playback(custom/callmenow-announce)
same =n,Verbose(2,end of macro)
Mitch
On 10/04/2012 02:30 PM, Ioan Indreias wrote:
Hello Mitch,
Hoping that the Queue application is not automatically Answering the
line (till an agent will do this) my suggestion is to switch between
who have to answer in order to progress to the second call leg. This
means that the Queue will be called through a Local Channel and the call
to your customer will be made through a Dial application.
Below is something to start with - in case it will work you could modify
to your needs.
[demo]
exten = s,1,NoOp(Queue without answer)
exten = s,2,Queue(sales)
Action: Originate
Channel: Local/s@demo/n
Application: Dial
Data: SIP/voipms/customer_number
HTH,
Ioan Indreias
Modulo Consulting // www.modulo.ro http://www.modulo.ro
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Re: [asterisk-users] Call me now outbound calls in a queue
Perfect! Thank you.
Mitch
On 10/05/2012 01:07 PM, Ioan Indreias wrote:
Hi Mitch,
Glad that it works for you.
Regarding the CallerID I suggest to set some the variables before the
actual Dial.
Something like:
Action: Originate
Channel: Local/s@callmenow/n
Context: to-customer
Exten: s
Priority: 1
Async: true
Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/customer_number
Variable: OUTBOUND_CALLERID_NUM=777
Variable: OUTBOUND_CALLERID_NAME=Call Me Now
Timeout: 99
[callmenow]
exten = s,1,NoOp(callmenow: Queue without answer)
same =n,Queue(sales,tc)
[dial-to-customer]
exten = s,1,NoOp(to-customer)
same = n,Set(CALLERID(num)=${__OUTBOUND_CALLERID_NUM})
same = n,Set(CALLERID(name)=${__OUTBOUND_CALLERID_NAME})
same = n,Dial(${CHANNEL_TO_CUSTOMER})
Have a nice weekend,
Ioan
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[asterisk-users] Parameterize asterisk config files
Asterisk 1.8 on Ubuntu We store the configuration files in CVS. We have a development, QA and production environments. 90% of the config files are the same across all 3 environments, but there are some differences in sip.conf and extensions.conf (environment specific voip providers and/or analog/digital lines). I'd like to be able to use the same config files in CVS and have the differences resolved at run time, based on host name of the asterisk server. Any ideas how to do this? I looked at STS, but it appears to be Mac only. One idea would be to use something like #include sip-$$$hostname$$$.conf and then use sed or similar in the startup script to replace $$$hostname$$$ with the actual host name. Then each host/environment would have it's own include file as needed. Another idea would be to write a simple perl or other program to pre-process the files and put some markers in the files themselves. ; onlyif host=abc ; /onlyif The pre-processor would delete lines between the tags that didn't match the currently running host. -- Mitch -- _ -- 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] Who said asterisk is not to the task
Sam - can you send output from a top when your server is under load? Just curious. -- Mitch -- _ -- 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] Call me now outbound calls in a queue
I want to put a call me now button on the web site that will place the request into an asterisk call queue and then when an agent picks up the call in the queue, place the outbound call to the customer. The following AMI command works, but it calls the customer first, before an agent is necessarily available. Action: Originate Channel: SIP/voipms/customer_number_here Context: external Async: true Application: Queue Data: sales Callerid: Company 8005551212 How can I get an available agent before the customer call is placed? -- _ -- 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
