Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]
Thank you Kevin, I've looked at your solution and while I agree it's not ideal it does appear to be something that might work for me. I'll see if I can maybe backport the QUEUE_MEMBER stuff to 1.8 from 11. I'm also exploring an idea with a co-worker of using an AMI listener that will fire off actions in response to the member being paused and doing things that way. I looked at parsing the log but sadly the log uses the Member Name in the log instead of the actual device so I don't have a way of knowing what handset they are logged into the queue from. On Wed, Mar 25, 2015 at 12:13 PM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote: > > First, let me say I feel dirty for even posting this. It is probably far > from ideal, but it does get the job done. I had the same issue. Also, I am > using Asterisk 11. I just looked and it doesn't appear that the > QUEUE_MEMBER function supports the paused option in 1.8. To be honest, I am > not sure if there is a good replacement for what I have done below in the > 1.8 series. > > It isn't elegant and if you have a lot of queues/queue members to check, > it will constitute a lot of looping, but it does work. Like you, I would > like to have a way to check the pause status of a member easier. If the > queue application could call a subroutine with it autopaused someone, that > would actually make an elegant solution, but for now, this was the way I > could see to do it. > > You could maybe call a script that would parse the queue_log file looking > for an agents status and pass that back into the dialplan. > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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 Quality Measuring
Hi Markus, Sounds interesting to me too... However my google-fu is letting me down today - I found VOIPmonitor at Sourceforge http://sourceforge.net/projects/voipmonitor/ but this looks like you'll need a license. Any chance you have a link to voipmon? Cheers .. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map) www.OntheNet.com.au NOTICE: This e-mail and any attachments are private and confidential and may contain privileged information. If you are not an authorised recipient, the copying or distribution of this e-mail and any attachments is prohibited and you must not read, print or act in reliance on this e-mail or attachments. Any pricing information supplied via email is an estimate or indicative only and may require a formal quotation to verify full terms and conditions. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler Sent: Thursday, 26 March 2015 7:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Quality Measuring Hi Patrick, try voipmon, there it's free and you can even track MOS. Markus Am 25.03.2015 um 14:21 schrieb Patrick Beaumont: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most > of the time it turns out to be their own broadband. Very occasionally > server load. Does anyone have any advice or links to advice on > measuring call quality? > > I’ve been playing around with “sip show channelstats” but can’t other > than measuring the packet loss I don’t really know what I’m supposed > to be looking for in order to say “ah ha! that’s the problem!”. I also > don’t know what it’s limits are. Will the stats in “sip show > channelstats” show a customer using a torrent client and saturating > their own broadband connection? > > Regards, > Patrick. > -- _ -- 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] Call Quality Measuring
Hi Patrick, try voipmon, there it's free and you can even track MOS. Markus Am 25.03.2015 um 14:21 schrieb Patrick Beaumont: Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I’ve been playing around with “sip show channelstats” but can’t other than measuring the packet loss I don’t really know what I’m supposed to be looking for in order to say “ah ha! that’s the problem!”. I also don’t know what it’s limits are. Will the stats in “sip show channelstats” show a customer using a torrent client and saturating their own broadband connection? Regards, Patrick. -- _ -- 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 13. Writing call quality parameters to CDR. How?
On Thu, 19 Mar 2015 10:12:22 +0100
Marek Cervenka wrote:
> because of problems you are facing i decided to go way with second table
>
> CREATE TABLE `cdr_extended` (
>`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
>`uniqueid` varchar(32) NOT NULL DEFAULT '',
> `callid` varchar(256) NOT NULL DEFAULT '' COMMENT 'sip call-id',
>`hangupcause` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci
> NOT NULL COMMENT 'info about hangup',
>`peerip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
>`recvip` varchar(15) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
>`from_u` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
>`uri` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
>`useragent` varchar(30) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT
> NULL,
>`codec1` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
>`codec2` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL,
>`llp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
> COMMENT 'lost packets by local end',
>`rlp` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
> COMMENT 'lost packets by remote end ',
>`ljitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
> COMMENT 'the same for jitter ',
>`rjitt` varchar(10) CHARACTER SET utf8 COLLATE utf8_czech_ci NOT NULL
> COMMENT 'the same for jitter ',
>PRIMARY KEY (`id`),
>KEY `uniqueid` (`uniqueid`)
> ) ENGINE=InnoDB DEFAULT CHARSET=utf8;
>
> in hangup handler or h exten i will use func_odbc
> like
> insert into cdr_extended (uniqueid,hangupcause,peerip,...) values
> ('${UNIQUEID}',...);
Interesting approach.
But how to tell from a call going directly (directmedia) and a call with
asterisk in between??
In the last case, two bridged channels, how to collect the parameters from each
"leg" in the "h" extension?
Cheers
Ethy
>
>
> Dne 18.3.2015 v 20:37 Dmitriy Serov napsal(a):
> > Hello.
> >
> > Voice quality when calling - this is one of the most important in the
> > PBX.
> > You need to record the quality parameters for each call to improve.
> >
> > Because the overall quality of a call can only be determined upon
> > completion, I did it in the HangUp handler and wrote in custom fields
> > of CDR.
> > This worked well in asterisk 11.
> >
> > In asterisk 13 I did not find a handler after the call, but before
> > finalizing the CDR.
> > I tried to call the AGI and there to update the CDR record by unique
> > identifiers. But faced with the fact that there are no needed record
> > in the table yet.
> > To write the data into a separate table and join them may be an
> > option. But do not want to resort to such a decision
> >
> > How do you solve this problem?
> >
> > Dmitriy Serov.
> >
>
>
> --
> ---
> Marek Cervenka
> ===
>
>
> --
> _
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--
Ethy H. Brito /"\
InterNexo Ltda. \ / CAMPANHA DA FITA ASCII - CONTRA MAIL HTML
+55 (12) 3797-6860 X ASCII RIBBON CAMPAIGN - AGAINST HTML MAIL
S.J.Campos - Brasil / \
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Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]
asterisk-users-boun...@lists.digium.com wrote on 03/25/2015 01:38:26 PM:
> I'm looking at enabling autopause on one of my queues where my queue
> members are bad about leaving their desks without pausing.
> The problem I see is that when the queue pauses an Member it doesn't
> jump into the dialplan to do so which means my handy device state
> and asterisk database driven Light for the Member showing their
> paused status won't update.
> My idea for solving this problem is to check the status of my Member
> in the queue before I send the calls into it and toggle on the
> Members Paused light at that point in time if they are paused.
> Sadly I don't see a way to determine if my Staff are paused or not
> from the dialplan, There doesn't appear to be a function to retrieve
> the status of the members in the queue.
> Does the list have any suggestions?
First, let me say I feel dirty for even posting this. It is probably far
from ideal, but it does get the job done. I had the same issue. Also, I am
using Asterisk 11. I just looked and it doesn't appear that the
QUEUE_MEMBER function supports the paused option in 1.8. To be honest, I
am not sure if there is a good replacement for what I have done below in
the 1.8 series.
[sub_autopause_status]
exten => s,1,NoOp(Checking for autopaused members for ${arg1} queue)
same => n,Set(MEMBERS=${QUEUE_MEMBER_LIST(${arg1})})
same => n,Set(i=1)
same => n,Set(max=${FIELDQTY(MEMBERS,,)})
same => n,While($[${i} <= ${max}])
same => n,Set(MEMBER=${CUT(MEMBERS,\,,${i})})
same => n,Set(STATUS=${QUEUE_MEMBER(${arg1},paused,${MEMBER})})
same => n,Set(MEMBER_EXT=${CUT(MEMBER,\/,2)})
same => n,ExecIf($["${STATUS}" = "0"]?System(echo "IN" >
/var/spool/asterisk/status/agent-${MEMBER_EXT}-status.txt))
same => n,ExecIf($["${STATUS}" = "1"]?System(echo "PAU" >
/var/spool/asterisk/status/agent-${MEMBER_EXT}-status.txt))
same => n,NoOp(${MEMBER}: ${STATUS})
same => n,Set(i=$[${i} + 1])
same => n,EndWhile()
same => n,Return()
So, as an explanation, I have multiple queues and agents who autopause. I
show their status on their phones, hence the System(echo...) commands to
the /var/spool/asterisk/status directory. Those files are used to generate
a simple web page that is shown on their phones that lets them see their
status. You should be able to adapt that to what you do.
Basically, you pass the queue name into the subroutine as arg1. The
subroutine gets a list of every person logged into that queue and then
loops through checking the status of each person using the QUEUE_MEMBER
function.
It isn't elegant and if you have a lot of queues/queue members to check,
it will constitute a lot of looping, but it does work. Like you, I would
like to have a way to check the pause status of a member easier. If the
queue application could call a subroutine with it autopaused someone, that
would actually make an elegant solution, but for now, this was the way I
could see to do it.
You could maybe call a script that would parse the queue_log file looking
for an agents status and pass that back into the dialplan.--
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[asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]
Howdy, I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan to do so which means my handy device state and asterisk database driven Light for the Member showing their paused status won't update. My idea for solving this problem is to check the status of my Member in the queue before I send the calls into it and toggle on the Members Paused light at that point in time if they are paused. Sadly I don't see a way to determine if my Staff are paused or not from the dialplan, There doesn't appear to be a function to retrieve the status of the members in the queue. Does the list have any suggestions? -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -- _ -- 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] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling
Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's internal extensions via the DID (and appropriate dialplans) but I am not able to make outbound calls to the PSTN from my (internal) extensions. I have the appropriate dialplans and I know the Asterisk server is getting in touch with the SIP.US server (see http://lists.digium.com/pipermail/asterisk-users/2015-March/286176.html which is the error I get). My question is, does anybody have a working pjsip.conf with SIP.US I could use? It has to be pjsip.conf (and not the wizard based configuration introduced in 13.2.0). Do I need to set up an outbound_proxy for SIP.US? Any help is deeply appreciated. Thank you! Alternately, could you help me with my config (a copy is below, changed some sensitive fields for obvious reasons)? I have configured my trunks in the following manner (based on https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples, and other pages on the same wiki, but there are small changes between them which confused the heck out of me): [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 local_net=192.168.1.0/24 external_media_address=aa.bb.cc.dd ; replaced real public IP address external_signaling_address=aa.bb.cc.dd ; replaced real public IP address [sonnyGW1] type=registration transport=transport-udp outbound_auth=sonnyGW1_auth server_uri=sip:regist...@gw1.sip.us ; no registrar@ in URI client_uri=sip:so...@gw1.sip.us contact_user=16175551212 ; replaced real DID retry_interval=60 forbidden_retry_interval=600 expiration=3600 [sonnyGW1_auth] type=auth auth_type=userpass password=** username=sonny ;realm=65.254.44.194 ;realm=gw1.sip.us [sonnyGW1] type=aor contact=sip:sonnyGW1@65.254.44.194:5060 ; tried also no username in URI [sonnyGW1] type=endpoint transport=transport-udp context=fromgw allow=!all,ulaw outbound_auth=sonnyGW1_auth aors=sonnyGW1 from_domain=gw1.sip.us [sonnyGW1] type=identify endpoint=sonnyGW1 match=65.254.44.194 ;; All endpoints for internal extensions follow -- _ -- 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 Quality Measuring (Laszlo)
Have you tried using tcpdump? Then analyze the pcap on wireshark? Marlon Araujo > On Mar 25, 2015, at 13:00, asterisk-users-requ...@lists.digium.com wrote: > > 1. Re: Call Quality Measuring (Laszlo) -- _ -- 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] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
thank you for your response but i think that the issue is related to the
RTP because i can call all numbers with the same format
when i call any number except 0033149xx i get the same adress from
provider only with this number cnfigurerd in ip-phone in our network i get
this error
best regards
number works without issue
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033661223291
-- SIP/FD-011f is making progress passing it to SIP/306-011e
> 0x2afee8182fa0 -- Probation passed - setting RTP source address to
192.168.1.212:12728 ip adress of my x-lite
> 0x2afee822e480 -- Probation passed - setting RTP source address to
217.195.31.148:43486ip adress of provider
SIP/FD-011f answered SIP/306-011e
> 0x2afee822e480 -- Probation passed - setting RTP source address to
217.195.31.148:43486 the same ip adress and the same port
number with error
Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
- Called SIP/FD/0033149xx
SIP/FD-011d is making progress passing it to SIP/306-011c
> 0x2afee8182fa0 -- Probation passed - setting RTP source address to
192.168.1.212:47452ip adress of my x-lite
> 0xc7452e0 -- Probation passed - setting RTP source address to
217.195.31.146:23392ip adress of provider
Got SIP response 556 "No address found" back from 217.195.31.129:5060
not the same ip and port
2015-03-25 13:47 GMT+00:00 A J Stiles :
> ** THIS IS NOT WHERE YOUR REPLY BELONGS **
>
> On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
> > tnaks for your response but the number dialed exist and i can call this
> > number when i configure the trunk directly in x-lite and i call call also
> > this number from my cell phone .
> > any help
> > thanks and regards
>
> Make sure you are sending the number in the correct format, when you Dial()
> via your trunk. Some providers want you to omit the leading zero from the
> STD
> code. Others want you to include it. Others still want you to include the
> IDD code (and then definitely leave out the 0, just like you were phoning
> home
> from abroad).
>
> My home phone number is (01332) XX. To call it, you might have to
> Dial()
> any of the following (assuming OUTSIDE is defined elsewhere):
>
> Dial(${OUTSIDE}/01332XX, 60); with leading 0
> Dial(${OUTSIDE}/1332XX, 60) ; without leading 0
> Dial(${OUTSIDE}/441332XX, 60) ; with IDD code
>
> If you don't know what format your telco are expecting and have to
> determine
> by experiment, it probably would be easiest to set up an extension which
> just
> makes a call to one fixed number -- your own mobile is as good as anything
> else.
>
> To remove the leading 0 from ${EXTEN} , you can use ${EXTEN:1} which omits
> one
> digit from the beginning.
>
> --
> AJS
>
> Note: Originating address only accepts e-mail from list! If replying off-
> list, change address to asterisk1list at earthshod dot co dot uk .
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Call Quality Measuring
On Wed, Mar 25, 2015 at 2:21 PM, Patrick Beaumont < p.beaum...@hatsoffsoftware.co.uk> wrote: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have any advice or links to advice on measuring call > quality? > > I’ve been playing around with “sip show channelstats” but can’t other than > measuring the packet loss I don’t really know what I’m supposed to be > looking for in order to say “ah ha! that’s the problem!”. I also don’t > know what it’s limits are. Will the stats in “sip show channelstats” show > a customer using a torrent client and saturating their own broadband > connection? > > Regards, > Patrick. > > -- > _ > -- 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 You can try voipmonitor (http://voipmonitor.org) free for 30 days, hopefully it's enough for finding and fixing the call quality issues. (I'm not affiliated with voipmonitor) -- -- Kind regards, Laszlo Bekesi http://voipfreak.net -- _ -- 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] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
** THIS IS NOT WHERE YOUR REPLY BELONGS **
On Wednesday 25 Mar 2015, Salaheddine Elharit wrote:
> tnaks for your response but the number dialed exist and i can call this
> number when i configure the trunk directly in x-lite and i call call also
> this number from my cell phone .
> any help
> thanks and regards
Make sure you are sending the number in the correct format, when you Dial()
via your trunk. Some providers want you to omit the leading zero from the STD
code. Others want you to include it. Others still want you to include the
IDD code (and then definitely leave out the 0, just like you were phoning home
from abroad).
My home phone number is (01332) XX. To call it, you might have to Dial()
any of the following (assuming OUTSIDE is defined elsewhere):
Dial(${OUTSIDE}/01332XX, 60); with leading 0
Dial(${OUTSIDE}/1332XX, 60) ; without leading 0
Dial(${OUTSIDE}/441332XX, 60) ; with IDD code
If you don't know what format your telco are expecting and have to determine
by experiment, it probably would be easiest to set up an extension which just
makes a call to one fixed number -- your own mobile is as good as anything
else.
To remove the leading 0 from ${EXTEN} , you can use ${EXTEN:1} which omits one
digit from the beginning.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
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Re: [asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
tnaks for your response but the number dialed exist and i can call this
number when i configure the trunk directly in x-lite and i call call also
this number from my cell phone .
any help
thanks and regards
2015-03-25 12:59 GMT+00:00 Matthew Jordan :
> On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
> wrote:
> > hello list,
> >
> > i have asterisk 11.15.0 and i have some trunks sip from my provider
> >
> > we have some ip phone astra 6731i
> >
> > each Ip-phone is configured with trunk and we call
> >
> > no ihave configured another trunk from the same provider in my asterisk
> >
> > i can call all numbers just the numbers are configured in thses ip
> phones.
> >
> > but when i configured the same trunk in x-lite i can call theses
> ip-phones
> > without issue
> > the problem just when i configure the trunk in my server and i use
> > extension
> >
> > all the ip-phone and x-lite and server asterisk in the same network
> > 192.168.1.x
> >
> > == Using SIP RTP TOS bits 184
> > == Using SIP RTP CoS mark 5
> > -- Called SIP/FD/0033149XX
> > -- SIP/FD-00b9 is making progress passing it to SIP/306-00b8
> >> 0x2afec424c430 -- Probation passed - setting RTP source address
> to
> > 192.168.1.212:57592
> >> 0xc5922b0 -- Probation passed - setting RTP source address to
> > 217.195.xx.xxx:29674
> > -- Got SIP response 556 "No address found" back from
> 217.195.XX.XXX:5060
> > == Everyone is busy/congested at this time (1:0/1/0)
> > -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/306-00b8",
> "Dial
> > failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE =
> 34")
> > in new stack
> > -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/306-00b8",
> > "0?continue,1:s-CONGESTION,1") in new stack
> > -- Goto (macro-dialout-trunk,s-CONGESTION,1)
> > -- Executing [s-CONGESTION@macro-dialout-trunk:1]
> > Set("SIP/306-00b8", "RC=34") in new stack
> > -- Executing [s-CONGESTION@macro-dialout-trunk:2]
> > Goto("SIP/306-00b8", "34,1") in new stack
> > -- Goto (macro-dialout-trunk,34,1)
> > -- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-00b8",
> > "continue,1") in new stack
> > -- Goto (macro-dialout-trunk,continue,1)
> > -- Executing [continue@macro-dialout-trunk:1]
> NoOp("SIP/306-00b8",
> > "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to
> > other trunks") in new stack
> > -- Executing [continue@macro-dialout-trunk:2]
> Set("SIP/306-00b8",
> > "CALLERID(number)=306") in new stack
> > -- Executing [0149XX@from-internal:7] Macro("SIP/306-00b8",
> > "outisbusy,") in new stack
> > -- Executing [s@macro-outisbusy:1] Progress("SIP/306-00b8", "")
> in
> > new stack
> > -- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-00b8",
> > "0?emergency,1") in new stack
> > -- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-00b8",
> > "0?intracompany,1") in new stack
> > -- Executing [s@macro-outisbusy:4] Playback("SIP/306-00b8",
> > "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
> > ast_openstream_full: File all-circuits-busy-now does not exist in any
> format
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
> > ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No
> > such file or directory
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
> > playback_exec: ast_streamfile failed on SIP/306-00b8 for
> > all-circuits-busy-now&pls-try-call-later, noanswer
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
> > ast_openstream_full: File pls-try-call-later does not exist in any format
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
> > ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No
> such
> > file or directory
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
> > playback_exec: ast_streamfile failed on SIP/306-00b8 for
> > all-circuits-busy-now&pls-try-call-later, noanswer
> > -- Executing [s@macro-outisbusy:5] Congestion("SIP/306-00b8",
> "20")
> > in new stack
> > [2015-03-25 12:18:31] WARNING[25161][C-006d]: channel.c:4862
> ast_prod:
> > Prodding channel 'SIP/306-00b8' failed
> > == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
> > 'SIP/306-00b8' in macro 'outisbusy'
> > == Spawn extension (from-internal, 0149XX, 7) exited non-zero on
> > 'SIP/306-00b8'
> > -- Executing [h@from-internal:1] Hangup("SIP/306-00b8", "") in
> new
> > stack
> > == Spawn extension (from-internal, h, 1) exited non-zero on
> > 'SIP/306-00b8'
> > == MixMonitor close filestream (mixed)
> > == End MixMonitor Recording SIP/306-00b8
> >
>
> The verbose output states why your call is congested:
>
> -- Got SIP response 556 "No address found" back from
> 217.195.XX.XXX:5
[asterisk-users] Call Quality Measuring
Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I’ve been playing around with “sip show channelstats” but can’t other than measuring the packet loss I don’t really know what I’m supposed to be looking for in order to say “ah ha! that’s the problem!”. I also don’t know what it’s limits are. Will the stats in “sip show channelstats” show a customer using a torrent client and saturating their own broadband connection? Regards, Patrick. -- _ -- 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] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
On Wed, Mar 25, 2015 at 7:35 AM, Salaheddine Elharit
wrote:
> hello list,
>
> i have asterisk 11.15.0 and i have some trunks sip from my provider
>
> we have some ip phone astra 6731i
>
> each Ip-phone is configured with trunk and we call
>
> no ihave configured another trunk from the same provider in my asterisk
>
> i can call all numbers just the numbers are configured in thses ip phones.
>
> but when i configured the same trunk in x-lite i can call theses ip-phones
> without issue
> the problem just when i configure the trunk in my server and i use
> extension
>
> all the ip-phone and x-lite and server asterisk in the same network
> 192.168.1.x
>
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> -- Called SIP/FD/0033149XX
> -- SIP/FD-00b9 is making progress passing it to SIP/306-00b8
>> 0x2afec424c430 -- Probation passed - setting RTP source address to
> 192.168.1.212:57592
>> 0xc5922b0 -- Probation passed - setting RTP source address to
> 217.195.xx.xxx:29674
> -- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060
> == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing [s@macro-dialout-trunk:23] NoOp("SIP/306-00b8", "Dial
> failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")
> in new stack
> -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/306-00b8",
> "0?continue,1:s-CONGESTION,1") in new stack
> -- Goto (macro-dialout-trunk,s-CONGESTION,1)
> -- Executing [s-CONGESTION@macro-dialout-trunk:1]
> Set("SIP/306-00b8", "RC=34") in new stack
> -- Executing [s-CONGESTION@macro-dialout-trunk:2]
> Goto("SIP/306-00b8", "34,1") in new stack
> -- Goto (macro-dialout-trunk,34,1)
> -- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-00b8",
> "continue,1") in new stack
> -- Goto (macro-dialout-trunk,continue,1)
> -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/306-00b8",
> "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to
> other trunks") in new stack
> -- Executing [continue@macro-dialout-trunk:2] Set("SIP/306-00b8",
> "CALLERID(number)=306") in new stack
> -- Executing [0149XX@from-internal:7] Macro("SIP/306-00b8",
> "outisbusy,") in new stack
> -- Executing [s@macro-outisbusy:1] Progress("SIP/306-00b8", "") in
> new stack
> -- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-00b8",
> "0?emergency,1") in new stack
> -- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-00b8",
> "0?intracompany,1") in new stack
> -- Executing [s@macro-outisbusy:4] Playback("SIP/306-00b8",
> "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
> ast_openstream_full: File all-circuits-busy-now does not exist in any format
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
> ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No
> such file or directory
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
> playback_exec: ast_streamfile failed on SIP/306-00b8 for
> all-circuits-busy-now&pls-try-call-later, noanswer
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
> ast_openstream_full: File pls-try-call-later does not exist in any format
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
> ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such
> file or directory
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
> playback_exec: ast_streamfile failed on SIP/306-00b8 for
> all-circuits-busy-now&pls-try-call-later, noanswer
> -- Executing [s@macro-outisbusy:5] Congestion("SIP/306-00b8", "20")
> in new stack
> [2015-03-25 12:18:31] WARNING[25161][C-006d]: channel.c:4862 ast_prod:
> Prodding channel 'SIP/306-00b8' failed
> == Spawn extension (macro-outisbusy, s, 5) exited non-zero on
> 'SIP/306-00b8' in macro 'outisbusy'
> == Spawn extension (from-internal, 0149XX, 7) exited non-zero on
> 'SIP/306-00b8'
> -- Executing [h@from-internal:1] Hangup("SIP/306-00b8", "") in new
> stack
> == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/306-00b8'
> == MixMonitor close filestream (mixed)
> == End MixMonitor Recording SIP/306-00b8
>
The verbose output states why your call is congested:
-- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060
The far end came back with a 556 response to the outbound INVITE
request. It doesn't think that whatever you dialled exists.
--
Matthew Jordan
Digium, Inc. | Director of Technology
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
[asterisk-users] TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34
hello list,
i have asterisk 11.15.0 and i have some trunks sip from my provider
we have some ip phone astra 6731i
each Ip-phone is configured with trunk and we call
no ihave configured another trunk from the same provider in my asterisk
i can call all numbers just the numbers are configured in thses ip phones.
but when i configured the same trunk in x-lite i can call theses ip-phones
without issue
the problem just when i configure the trunk in my server and i use
extension
all the ip-phone and x-lite and server asterisk in the same network
192.168.1.x
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/FD/0033149XX
-- SIP/FD-00b9 is making progress passing it to SIP/306-00b8
> 0x2afec424c430 -- Probation passed - setting RTP source address to
192.168.1.212:57592
> 0xc5922b0 -- Probation passed - setting RTP source address to
217.195.xx.xxx:29674
-- Got SIP response 556 "No address found" back from 217.195.XX.XXX:5060
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/306-00b8", "Dial
failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34")
in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/306-00b8",
"0?continue,1:s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1]
Set("SIP/306-00b8", "RC=34") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2]
Goto("SIP/306-00b8", "34,1") in new stack
-- Goto (macro-dialout-trunk,34,1)
-- Executing [34@macro-dialout-trunk:1] Goto("SIP/306-00b8",
"continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/306-00b8",
"TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to
other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/306-00b8",
"CALLERID(number)=306") in new stack
-- Executing [0149XX@from-internal:7] Macro("SIP/306-00b8",
"outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/306-00b8", "") in
new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/306-00b8",
"0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/306-00b8",
"0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/306-00b8",
"all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
ast_openstream_full: File all-circuits-busy-now does not exist in any format
[2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
ast_streamfile: Unable to open all-circuits-busy-now (format (ulaw)): No
such file or directory
[2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
playback_exec: ast_streamfile failed on SIP/306-00b8 for
all-circuits-busy-now&pls-try-call-later, noanswer
[2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:701
ast_openstream_full: File pls-try-call-later does not exist in any format
[2015-03-25 12:18:31] WARNING[25161][C-006d]: file.c:1017
ast_streamfile: Unable to open pls-try-call-later (format (ulaw)): No such
file or directory
[2015-03-25 12:18:31] WARNING[25161][C-006d]: app_playback.c:484
playback_exec: ast_streamfile failed on SIP/306-00b8 for
all-circuits-busy-now&pls-try-call-later, noanswer
-- Executing [s@macro-outisbusy:5] Congestion("SIP/306-00b8", "20")
in new stack
[2015-03-25 12:18:31] WARNING[25161][C-006d]: channel.c:4862 ast_prod:
Prodding channel 'SIP/306-00b8' failed
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on
'SIP/306-00b8' in macro 'outisbusy'
== Spawn extension (from-internal, 0149XX, 7) exited non-zero on
'SIP/306-00b8'
-- Executing [h@from-internal:1] Hangup("SIP/306-00b8", "") in new
stack
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/306-00b8'
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/306-00b8
--
_
-- 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 11.14 - voicemail incorrect duration
Hi Stefan, Dominique Haeber schrieb am Die, 27. Jan 08:55: > I have looked at the time and talked for at least 4 seconds. > In CLI log are 5-6 seconds visible between open to writing and Hang > up. > Nevertheless, Asterisk writes about two seconds. > > > The value for silencethreshold (140) is unusually large. > > It would be worth a try to set the value down. > In asterisk 1.6 this value was still good. But that is far back > again... > I will write again. This was the solution. Thank you! Greetings Dominique -- _ -- 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
