Re: [asterisk-users] Asterisk 11 -CDR values changed in hangup handler not saved ?
check out the endbeforehexten option in cdr.conf this needs to set to yes Julian On 28 March 2013 23:56, Olivier oza_4...@yahoo.fr 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
2013/3/29 Julian Lyndon-Smith aster...@dotr.com check out the endbeforehexten option in cdr.conf this needs to set to yes Julian Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the saving to file part. Then my issue is I can't update CDR value is hangup exten. Here is a dialplan that illustrate this: [from-foobar] exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, Set(CDR(userfield)=foo) same = n, Dial(SIP/foobar/${EXTEN}) same = n, Set(CDR(userfield)=bar) same = n, Hangup() exten = h,1,Verbose(0,Entering context ${CONTEXT} from ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2) My goal is to get either baz1 or baz2 value in /var/log/asterisk/cdr-csv/Master.csv. Typing channel originate Local/7005@from-foobar application Playback tt-monkeys, I can see that the line with ExecIf is run but CDR still contains foo value (the one set before Dial). The strange thing is : 1. a CDR is written at the moment extension 7005 answers, 2. no other CDR is added when 7005 hangs up (so can't tell how long extension 7005 listened to monkeys fellows). (Setting endbeforehexten to either yes or no has no effect on this behaviour. My question are: 1. Is it simply possible to update CDR in hangup exten ? 2. How can I have a CDR for the application Playback part (see above) ? 3. Any tip or suggestion ? Cheers -- _ -- 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 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
Ah, right. Have a look at this documentation: You may need to add some mapping Julian cdr_custom This CDR backend allows for custom formatting of CDR records in a log file. This module is most commonly used for customized CSV output. The configuration file used for this module is /etc/asterisk/cdr_custom.conf. A single section called [mappings] should exist in this file. The [mappings] section contains mappings between a filename and the custom template for a CDR. The template is specified using Asterisk dialplan functions. The following example shows a sample configuration for cdr_custom that enables a single CDR log file, Master.csv. This file will be created as /var/log/asterisk/cdr-custom/Master.csv. The template that has been defined uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function retrieves values from the CDR being logged. The CSV_QUOTE() function ensures that the values are properly escaped for the CSV file format: [mappings] Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})}, ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})}, ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})}, ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})}, ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})}, ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})}, ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})}, ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})}, ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})} In the actual configuration file, the value in the Master.csv mapping should be on a single line. cdr_manager On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote: 2013/3/29 Julian Lyndon-Smith aster...@dotr.com check out the endbeforehexten option in cdr.conf this needs to set to yes Julian Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the saving to file part. Then my issue is I can't update CDR value is hangup exten. Here is a dialplan that illustrate this: [from-foobar] exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, Set(CDR(userfield)=foo) same = n, Dial(SIP/foobar/${EXTEN}) same = n, Set(CDR(userfield)=bar) same = n, Hangup() exten = h,1,Verbose(0,Entering context ${CONTEXT} from ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2) My goal is to get either baz1 or baz2 value in /var/log/asterisk/cdr-csv/Master.csv. Typing channel originate Local/7005@from-foobar application Playback tt-monkeys, I can see that the line with ExecIf is run but CDR still contains foo value (the one set before Dial). The strange thing is : 1. a CDR is written at the moment extension 7005 answers, 2. no other CDR is added when 7005 hangs up (so can't tell how long extension 7005 listened to monkeys fellows). (Setting endbeforehexten to either yes or no has no effect on this behaviour. My question are: 1. Is it simply possible to update CDR in hangup exten ? 2. How can I have a CDR for the application Playback part (see above) ? 3. Any tip or suggestion ? Cheers -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] sip set debug on output to file only (not to console)
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, -- marie -- _ -- 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)
Marie Fischer wrote: full = notice,warning,error,debug,verbose,dtmf,fax You should have a log called full in: /var/log/asterisk Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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)
Marie Fischer wrote: but the SIP log clutters up the console quite badly I guess I should slow down when reading. Sorry for the noise. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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)
On 29.03.2013, at 15:05, Doug Lytle supp...@drdos.info wrote: Marie Fischer wrote: full = notice,warning,error,debug,verbose,dtmf,fax You should have a log called full in: /var/log/asterisk Sure I do and happy with that. :) The point is, I also have my Asterisk console full of SIP messages and I asked if it was possible to switch those off. -- marie -- _ -- 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 ?
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)
hi, open debug only on problematic peer. sip set debug peer peer name or sip set debug ip peer ip On Fri, Mar 29, 2013 at 2:02 PM, Marie Fischer ma...@vtl.ee 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, -- marie -- _ -- 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 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
Thanks but I willingly choose a standard CDR field (I checked with both accountcode and userfield) which appears in /var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv away to simplify things) the fact found in Master.csv is foo, the value set before entering the hangup extension (see previous dialplan)). To me, this is either a feature (you can't set CDR values in hangup exten) or a bug. How would you qualify this ? 2013/3/29 Julian Lyndon-Smith aster...@dotr.com Ah, right. Have a look at this documentation: You may need to add some mapping Julian cdr_custom This CDR backend allows for custom formatting of CDR records in a log file. This module is most commonly used for customized CSV output. The configuration file used for this module is /etc/asterisk/cdr_custom.conf. A single section called [mappings] should exist in this file. The [mappings] section contains mappings between a filename and the custom template for a CDR. The template is specified using Asterisk dialplan functions. The following example shows a sample configuration for cdr_custom that enables a single CDR log file, Master.csv. This file will be created as /var/log/asterisk/cdr-custom/Master.csv. The template that has been defined uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function retrieves values from the CDR being logged. The CSV_QUOTE() function ensures that the values are properly escaped for the CSV file format: [mappings] Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})}, ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})}, ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})}, ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})}, ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})}, ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})}, ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})}, ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})}, ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})} In the actual configuration file, the value in the Master.csv mapping should be on a single line. cdr_manager On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote: 2013/3/29 Julian Lyndon-Smith aster...@dotr.com check out the endbeforehexten option in cdr.conf this needs to set to yes Julian Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the saving to file part. Then my issue is I can't update CDR value is hangup exten. Here is a dialplan that illustrate this: [from-foobar] exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, Set(CDR(userfield)=foo) same = n, Dial(SIP/foobar/${EXTEN}) same = n, Set(CDR(userfield)=bar) same = n, Hangup() exten = h,1,Verbose(0,Entering context ${CONTEXT} from ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2) My goal is to get either baz1 or baz2 value in /var/log/asterisk/cdr-csv/Master.csv. Typing channel originate Local/7005@from-foobar application Playback tt-monkeys, I can see that the line with ExecIf is run but CDR still contains foo value (the one set before Dial). The strange thing is : 1. a CDR is written at the moment extension 7005 answers, 2. no other CDR is added when 7005 hangs up (so can't tell how long extension 7005 listened to monkeys fellows). (Setting endbeforehexten to either yes or no has no effect on this behaviour. My question are: 1. Is it simply possible to update CDR in hangup exten ? 2. How can I have a CDR for the application Playback part (see above) ? 3. Any tip or suggestion ? Cheers -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- 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] To queue or not to queue...
Hello Gregory, I wouldn't say this is a typical scenario for using a ringall queue, especially if the agent set gets larger and larger. On the other side, a ringgroup won't solve the issue of ringing all those phones at once. What I would be looking into, considered the motivation of your agents, is to split the system into more than one queue and send the calls randomly to each queue. If everybody is busy you get out and retry. This should not impact call answer times as long as you have 30/40 people available per queue - but your box will handle a fraction of the load and you can easily partition such a system on multiple boxes. Just my two cents, l. 2013/3/28 Gregory Malsack gmals...@coastalacq.com Hello All, History ~ I recently took a position with a call center. At the time they had about 50 agents in a call queue. The queue was setup to ringall. The agents use Eyebeam softphones. Everything is local lan, no routers, everything connected via Cisco 3600 10/100 switches. Now we are up to about 150 agents, and I have kept everything pretty much the same way for a couple of reasons. However, those reasons are slowly drifting away and it's become the right time for me to start questioning some of the previous configuration. Here's the scenario~ 150 agents, all are commission based sales reps. 99% of the calls are answered within the first ring. the rest are answered between the second and third ring. Never in my 4 months with the company has a queue call been in the queue more then 20 seconds. Problem~ Several times a week or sometimes a day, the reps will tell me that the same call will be answered by 3 or 4 or 5 reps, and none of them get the inbound audio. Asterisk only shows 1 of the reps actually connecting the call, however the call logs in Eyebeam for all 5 reps, show that they took the call and were connected for a short period of time before disconnecting the call because there is no inbound audio. Point of discussion~ Is there really a reason to maintain a queue? With the companies growth they are now discussing the option of sending certain affiliates to certain sales reps. Am I better off using ring groups? Additionally I am working towards running as much of my configs via mysql as possible and turning up multiple servers to handle the calls. So far we have reached 130 simultaneous calls on one server, and about 10,000 calls processed during a 12 hour day. Thanks for reading. I look forward to hearing peoples views on this... -- _ -- 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 -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- 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 ?
How would you qualify it ? A feature ? A bug ? Could you find a work around ? 2013/3/29 Mitch Claborn 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-**usershttp://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-**usershttp://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] Getting Unknown Error while configuring Asterisk with Linux HA
Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include monitor_sipuri= sip:42@10.3.152.103 parameter in primitive section it is giving me an errors like listed below; root@asterisk2 ~ crm_mon -1 Last updated: Thu Mar 28 06:09:54 2013 Stack: Heartbeat Current DC: asterisk2 (b966dfa2-5973-4dfc-96ba-b2d38319c174) - partition with quorum Version: 1.0.12-unknown 2 Nodes configured, unknown expected votes 1 Resources configured. Online: [ asterisk1 asterisk2 ] Resource Group: group_1 asterisk_2 (lsb:asterisk): Started asterisk1 IPaddr_10_3_152_103(ocf::heartbeat:IPaddr):Started asterisk1 Failed actions: p_asterisk_start_0 (node=asterisk1, call=64, rc=1, status=complete): unknown error p_asterisk_start_0 (node=asterisk2, call=20, rc=1, status=complete): unknown error I tested the 'sipsak' tool on cli, it is executing without any issue i.e. returning 200 OK but when I remove this param monitor_sipuri I'm not getting the errors and also I created sip profile '42' without setting any password, tested first on softphone and is working. Test result for sipsak; root@asterisk1 ~ sipsak -v -s sip:42@10.3.152.103 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.3.152.105:60928 ;branch=z9hG4bK.274e15e9;alias;received=10.3.152.103;rport=60928 From: sip:sipsak@10.3.152.105:60928;tag=68c5c65d To: sip:42@10.3.152.103;tag=as558d9271 Call-ID: 1757791837@10.3.152.105 CSeq: 1 OPTIONS Server: Asterisk PBX 10.12.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:10.3.152.103:5060 Accept: application/sdp Content-Length: 0 Listing down the configuration below which I configured; node $id=887bae58-1eb6-47d1-b539-d12a2ed3d836 asterisk1 node $id=b966dfa2-5973-4dfc-96ba-b2d38319c174 asterisk2 primitive IPaddr_10_3_152_103 ocf:heartbeat:IPaddr \ op monitor interval=5s timeout=20s \ params ip=10.3.152.103 primitive p_asterisk ocf:heartbeat:asterisk \ op monitor interval=10s \ params realtime=true group group_1 p_asterisk IPaddr_10_3_152_103 \ meta target-role=Started location rsc_location_group_1 group_1 \ rule $id=preferred_location_group_1 100: #uname eq asterisk1 colocation asterisk-with-ip inf: p_asterisk IPaddr_10_3_152_103 property $id=cib-bootstrap-options \ symmetric-cluster=true \ no-quorum-policy=stop \ default-resource-stickiness=0 \ stonith-enabled=false \ stonith-action=reboot \ startup-fencing=true \ stop-orphan-resources=true \ stop-orphan-actions=true \ remove-after-stop=false \ default-action-timeout=120s \ is-managed-default=true \ cluster-delay=60s \ pe-error-series-max=-1 \ pe-warn-series-max=-1 \ pe-input-series-max=-1 \ dc-version=1.0.12-unknown \ cluster-infrastructure=Heartbeat And the status I'm getting is listed below; root@asterisk1 ~ crm_mon -1 Last updated: Fri Mar 29 12:25:10 2013 Stack: Heartbeat Current DC: asterisk1 (887bae58-1eb6-47d1-b539-d12a2ed3d836) - partition with quorum Version: 1.0.12-unknown 2 Nodes configured, unknown expected votes 1 Resources configured. Online: [ asterisk1 asterisk2 ] Resource Group: group_1 p_asterisk (ocf::heartbeat:asterisk): Started asterisk1 IPaddr_10_3_152_103(ocf::heartbeat:IPaddr):Started asterisk1 Please advise to overcome this issue. -- Regards, Ahmed Munir Chohan -- _ -- 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 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
you can't set values in the h extension _unless_ you use the endbeforehexten option in cdr.conf you may need to reload the cdr module or restart asterisk for the option to take effect. It works. I know it does, as I use hangup handlers all the time. Much better than the h extension ;) Julian On 29 March 2013 14:06, Olivier oza_4...@yahoo.fr wrote: Thanks but I willingly choose a standard CDR field (I checked with both accountcode and userfield) which appears in /var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv away to simplify things) the fact found in Master.csv is foo, the value set before entering the hangup extension (see previous dialplan)). To me, this is either a feature (you can't set CDR values in hangup exten) or a bug. How would you qualify this ? 2013/3/29 Julian Lyndon-Smith aster...@dotr.com Ah, right. Have a look at this documentation: You may need to add some mapping Julian cdr_custom This CDR backend allows for custom formatting of CDR records in a log file. This module is most commonly used for customized CSV output. The configuration file used for this module is /etc/asterisk/cdr_custom.conf. A single section called [mappings] should exist in this file. The [mappings] section contains mappings between a filename and the custom template for a CDR. The template is specified using Asterisk dialplan functions. The following example shows a sample configuration for cdr_custom that enables a single CDR log file, Master.csv. This file will be created as /var/log/asterisk/cdr-custom/Master.csv. The template that has been defined uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function retrieves values from the CDR being logged. The CSV_QUOTE() function ensures that the values are properly escaped for the CSV file format: [mappings] Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})}, ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})}, ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})}, ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})}, ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})}, ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})}, ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})}, ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})}, ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})} In the actual configuration file, the value in the Master.csv mapping should be on a single line. cdr_manager On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote: 2013/3/29 Julian Lyndon-Smith aster...@dotr.com check out the endbeforehexten option in cdr.conf this needs to set to yes Julian Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the saving to file part. Then my issue is I can't update CDR value is hangup exten. Here is a dialplan that illustrate this: [from-foobar] exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, Set(CDR(userfield)=foo) same = n, Dial(SIP/foobar/${EXTEN}) same = n, Set(CDR(userfield)=bar) same = n, Hangup() exten = h,1,Verbose(0,Entering context ${CONTEXT} from ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2) My goal is to get either baz1 or baz2 value in /var/log/asterisk/cdr-csv/Master.csv. Typing channel originate Local/7005@from-foobar application Playback tt-monkeys, I can see that the line with ExecIf is run but CDR still contains foo value (the one set before Dial). The strange thing is : 1. a CDR is written at the moment extension 7005 answers, 2. no other CDR is added when 7005 hangs up (so can't tell how long extension 7005 listened to monkeys fellows). (Setting endbeforehexten to either yes or no has no effect on this behaviour. My question are: 1. Is it simply possible to update CDR in hangup exten ? 2. How can I have a CDR for the application Playback part (see above) ? 3. Any tip or suggestion ? Cheers -- _ -- 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 -- Julian Lyndon-Smith IT Director, Dot R Limited I don’t care if it works on your machine! We are not shipping your machine!” The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg -- _ -- Bandwidth and Colocation
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] Asterisk 11 - Change CDR in hangup exten [Was: CDR values changed in hangup handler not saved]
Le 29 mars 2013 18:26, Julian Lyndon-Smith aster...@dotr.com a écrit : you can't set values in the h extension _unless_ you use the endbeforehexten option in cdr.conf I did and couldn't get it to work :-( you may need to reload the cdr module or restart asterisk for the option to take effect. It works. You mean it does work in asterisk 11 ? Which CDR variables did you then play with ? I know it does, as I use hangup handlers all the time. Much better than the h extension ;) I fully agree ! Julian On 29 March 2013 14:06, Olivier oza_4...@yahoo.fr wrote: Thanks but I willingly choose a standard CDR field (I checked with both accountcode and userfield) which appears in /var/log/asterisk/cdr-csv/Master.csv (to keep cdr-cusdom/Master.csv away to simplify things) the fact found in Master.csv is foo, the value set before entering the hangup extension (see previous dialplan)). To me, this is either a feature (you can't set CDR values in hangup exten) or a bug. How would you qualify this ? 2013/3/29 Julian Lyndon-Smith aster...@dotr.com Ah, right. Have a look at this documentation: You may need to add some mapping Julian cdr_custom This CDR backend allows for custom formatting of CDR records in a log file. This module is most commonly used for customized CSV output. The configuration file used for this module is /etc/asterisk/cdr_custom.conf. A single section called [mappings] should exist in this file. The [mappings] section contains mappings between a filename and the custom template for a CDR. The template is specified using Asterisk dialplan functions. The following example shows a sample configuration for cdr_custom that enables a single CDR log file, Master.csv. This file will be created as /var/log/asterisk/cdr-custom/Master.csv. The template that has been defined uses both the CDR() and CSV_QUOTE() dialplan functions. The CDR() function retrieves values from the CDR being logged. The CSV_QUOTE() function ensures that the values are properly escaped for the CSV file format: [mappings] Master.csv = ${CSV_QUOTE(${CDR(clid)})},${CSV_QUOTE(${CDR(src)})}, ${CSV_QUOTE(${CDR(dst)})},${CSV_QUOTE(${CDR(dcontext)})}, ${CSV_QUOTE(${CDR(channel)})},${CSV_QUOTE(${CDR(dstchannel)})}, ${CSV_QUOTE(${CDR(lastapp)})},${CSV_QUOTE(${CDR(lastdata)})}, ${CSV_QUOTE(${CDR(start)})},${CSV_QUOTE(${CDR(answer)})}, ${CSV_QUOTE(${CDR(end)})},${CSV_QUOTE(${CDR(duration)})}, ${CSV_QUOTE(${CDR(billsec)})},${CSV_QUOTE(${CDR(disposition)})}, ${CSV_QUOTE(${CDR(amaflags)})},${CSV_QUOTE(${CDR(accountcode)})}, ${CSV_QUOTE(${CDR(uniqueid)})},${CSV_QUOTE(${CDR(userfield)})} In the actual configuration file, the value in the Master.csv mapping should be on a single line. cdr_manager On 29 March 2013 10:02, Olivier oza_4...@yahoo.fr wrote: 2013/3/29 Julian Lyndon-Smith aster...@dotr.com check out the endbeforehexten option in cdr.conf this needs to set to yes Julian Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the saving to file part. Then my issue is I can't update CDR value is hangup exten. Here is a dialplan that illustrate this: [from-foobar] exten = _X.,1,Verbose(0,Entering context ${CONTEXT} from channel ${CHANNEL(channeltype)} ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, Set(CDR(userfield)=foo) same = n, Dial(SIP/foobar/${EXTEN}) same = n, Set(CDR(userfield)=bar) same = n, Hangup() exten = h,1,Verbose(0,Entering context ${CONTEXT} from ${CHANNEL(channeltype)} channel ${CHANNEL} with EXTEN and CID set to ${EXTEN} and ${CALLERID(num)}) same = n, ExecIf($[x${CHANNEL(channeltype)}=xLocal]?Set(CDR(userfield)=baz1:baz2) My goal is to get either baz1 or baz2 value in /var/log/asterisk/cdr-csv/Master.csv. Typing channel originate Local/7005@from-foobar application Playback tt-monkeys, I can see that the line with ExecIf is run but CDR still contains foo value (the one set before Dial). The strange thing is : 1. a CDR is written at the moment extension 7005 answers, 2. no other CDR is added when 7005 hangs up (so can't tell how long extension 7005 listened to monkeys fellows). (Setting endbeforehexten to either yes or no has no effect on this behaviour. My question are: 1. Is it simply possible to update CDR in hangup exten ? 2. How can I have a CDR for the application Playback part (see above) ? 3. Any tip or suggestion ? Cheers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Pattern matching repeating digits
Eric, Thanks; of course, this is also an option. However, setting up a separate context for this type of thing with several identical Goto statements also strikes me as inelegant, even if it is less so. -- Nathan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Thursday, March 28, 2013 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Pattern matching repeating digits You are correct, it is stupid 8-) exten = 233,1,Goto(dial-out,${EXTEN},1) exten = 255,1,Goto(dial-out,${EXTEN},1) [dial-out] exten = _XXX,1,DoStuff() exten = _XXX,n,AndMoreStuff() exten = _XXX,n,Dial(something) exten = _XXX,n,Hangup -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nathan Anderson Sent: Wednesday, March 27, 2013 2:18 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] Pattern matching repeating digits 'lo, all, Is there some (possibly undocumented?) way that I can pattern-match on a specified number of repeating digits? (Something similar to regular expressions' {}) Here's an example: let's say I have a string of things that need to be done for both extensions 233 and 255. I can either... A) Repeat the exact same code for both extensions, like so: exten = 233,1,DoStuff() exten = 233,n,AndMoreStuff() exten = 233,n,Dial(something) exten = 255,1,DoStuff() exten = 255,n,AndMoreStuff() exten = 255,n,Dial(something) ...which is stupid, or... B) I can attempt code reuse for similar cases (a Good Thing[tm]), and make as specific of a match as possible, like so: exten = _2[35][35],1,DoStuff() exten = _2[35][35],n,AndMoreStuff() exten = _2[35][35],n,Dial(something) ...but this will not only match 233 and 255, but 235 and 253 as well. It'd be nice if there was a substitute character that meant a character that is exactly the same as the preceding one; for example, if R was meant to represent such a concept, then this would do what I want: exten = _2[35]R,1,DoStuff() exten = _2[35]R,n,AndMoreStuff() exten = _2[35]R,n,Dial(something) You could even do crazy things like chain them together (this would match 2 and 2 and nothing else); exten = _2[35]RRR,1,DoStuff() exten = _2[35]RRR,n,AndMoreStuff() exten = _2[35]RRR,n,Dial(something) Am I missing something or does this really not exist? Thanks, -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- 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] IPv6
Hi all, I had to re-install a new machine and noticed that by default, ip was only listening on 0.0.0.0, thus ipv4 only. Easily changed. However, when looking at iax.conf, I found here the same, but it looks like iax is still ipv4 only? If i change bindaddr=192.168.0.1 towards bindaddr=::, and look with lsof -i iax is still not listening on V6. Is iax/ipv6 still on the TODO-list ? Hans -- _ -- 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] IPv6
On 03/29/2013 07:05 PM, Hans Witvliet wrote: Hi all, I had to re-install a new machine and noticed that by default, ip was only listening on 0.0.0.0, thus ipv4 only. Easily changed. However, when looking at iax.conf, I found here the same, but it looks like iax is still ipv4 only? If i change bindaddr=192.168.0.1 towards bindaddr=::, and look with lsof -i iax is still not listening on V6. Is iax/ipv6 still on the TODO-list ? The IAX2 channel driver does not currently support IPv6. While it would be a nice improvement, there are no plans to perform this work at this time. Of course, the Asterisk open source developer community could take this work on, which would be a welcome addition to Asterisk 12. Matt -- 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 -- _ -- 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