[asterisk-users] Calls running forever / CDRs inaccurate
Hi list! Running Asterisk 20.0.0 on CentOS 7, logging CDRs using cdr_adaptive_odbc to mariadb-server-5.5.68 (via mariadb-connector-odbc-3.1.7-ga-rhel7) Using chan_sip. I'm facing the problem when there is a sudden spike of calls, some of the calls that are being made during those spikes hang forever basically. This looks like this: [root@voip]# asterisk -rx 'core show channels verbose' |sort -r -k 9 | grep -v Outgoing Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode PeerAccount BridgeID SIP/customer01 customer-voipin 49xxx 26 RingDial SIP/+49xxx@provider4912:49:05 So, the longest currently active call is in state "Ringing" for 12 hours 49 minutes. It could be also that a call is in state "Up" for an arbitrary duration (10, 15, 20+ hours). A cron script is restarting Asterisk every night, and this is the moment these calls are cleared, otherwise they'd hang/run forever. Now, the problem is, that this is causing a billing discrepancy. I've had calls with billsec 20+ hours and state "Up" in the CDRs. The termination providers (there are multiple) are ending the calls after max. 120 minutes, so the issue is not there but in the local Asterisk. Any recommendations on how I could debug, or even fix, this? Maybe a workaround to help with the billing discrepancy? In theory, if I set TIMEOUT(absolute) to maybe 2 hours and 2 minutes, I could find and delete those ghost calls in the CDRs at the end of the month, before sending the invoice to the customer, but that's not a good solution either (because losing money). But maybe still better than overcharging the customer thousands of minutes?! :) This behavior happens every day on approx. 5-200 calls. (At the end of a day I could have 200 hanging/ghost calls). Help! :) Lastly, a "core show channel" on a hanging call, obfuscated: [root@voip asterisk]# asterisk -rx 'core show channel SIP/customer01-dfa4' -- General -- Name: SIP/customer01-dfa4 Type: SIP UniqueID: voip-1683277216.81923 LinkedID: voip-1683277216.81923 Caller ID: 49xx Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: 49x Language: en State: Ring (4) NativeFormats: (alaw) WriteFormat: alaw ReadFormat: alaw WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 13h4m11s Bridge ID: (Not bridged) -- PBX -- Context: customer-voipin Extension: 49x Priority: 26 Call Group: 0 Pickup Group: 0 Application: Dial Data: SIP/+49@provider Call Identifer: [C-85c3] Variables: PROGRESSTIME_MS= PROGRESSTIME= RINGTIME_MS= RINGTIME= DIALEDTIME_MS= DIALEDTIME= ANSWEREDTIME_MS= ANSWEREDTIME= DIALEDPEERNAME= DIALEDPEERNUMBER= DIALSTATUS= SIPADDHEADER02=X-Something: something AUTO_MONITOR=wav,/var/spool/asterisk/monitor/20230505110016-customer-DE-EXTEN-49xxx-CLINUM-49x-CLINAME--PAICLEAN--CLICLEAN-49x-OCLINUM--OCLINAME-,mX MONITOR_EXEC=/var/lib/asterisk/2wav2mp3.sh CALLFILENAME=20230505110016-customer-DE-EXTEN-49xxx-CLINUM-49-CLINAME--PAICLEAN--CLICLEAN-49xx-OCLINUM--OCLINAME- SIPADDHEADER01=P-Asserted-Identity: CLICLEAN=49xx CLILEN=12 SIPCALLID=85b9164eeb2211eda29c008cfa0447f8@x.x.x.x SIPDOMAIN=x.x.x.x SIPURI=sip:49x@x.x.x.x:5061 CDR Variables: level 1: customer=customer level 1: country=DE level 1: dnid=49 level 1: clid="" <49x> level 1: src=49 level 1: dst=49 level 1: dcontext=customer-voipin level 1: channel=SIP/customer01-dfa4 level 1: dstchannel=SIP/provider-dfa6 level 1: lastapp=Dial level 1: lastdata=SIP/+49@provider level 1: start=1683277216.169426 level 1: answer=0.00 level 1: end=0.00 level 1: duration=47051 level 1: billsec=0 level 1: disposition=1 level 1: amaflags=3 level 1: uniqueid=voip-1683277216.81923 level 1: linkedid=voip-1683277216.81923 level 1: sequence=64906 -- Streams -- Name: audio-0 Type: audio State: sendrecv0 Group: -1 Formats: (alaw) Metadata: Thank you! Markus -- _ -- 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] @-sign gets transmitted as %40 in outgoing SIP packets (CallerID)
Am 17.10.2022 um 15:07 schrieb Joshua C. Colp: The number is sent as the user portion in SIP. That can't have the "@" in it, as that is invalid and against spec - it gets turned into "%40". If what you ACTUALLY want is the user portion to be "anonymous" and the domain portion to be "anonymous.invalid" then you may be able to set the callerid presentation[1] to not be allowed: Set(CALLERID(pres)=prohib) Thank you very much, that was the solution! :) -- _ -- 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] @-sign gets transmitted as %40 in outgoing SIP packets (CallerID)
Hi list, I'm using Asterisk 11.25.0 and would like to set "anonymous@anonymous.invalid" as outgoing caller ID via SIP: Set(CALLERID(num)=anonymous@anonymous.invalid) However, when I look at the outgoing packet with tcpdump I see that the @ is not being transmitted correctly: From: ;tag=as26264e65 My question is how can I get Asterisk to send a "@" instead of "%40"? Thank you! Markus -- _ -- 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] Get a SHAKEN Identity Token
Am 07.01.2021 um 23:49 schrieb Alexander Perkins:
Hi All. We have old Asterisk servers, 1,89, (we cannot upgrade because
of several reasons) and we are now implementing SHAKEN via our
provider. We place a SIP call to our provider and they return a 302
(information below). I am trying to get the X-Identity information
below, but I do not seem to be able to do so. Can somebody help me with
this? Any suggestions on how to get it?
I use SIP_HEADER to extract information from inbound SIP packets and
SIPAddHeader to copy that info to the outgoing call leg. Maybe this
helps you?
Example:
exten => _+X.,1,NoOp(${CALLERID(num)})
exten => _+X.,n,Set(PAI=${SIP_HEADER(P-Asserted-Identity)})
exten => _+X.,n,Set(PAI=${CUT(PAI,:,2)})
exten => _+X.,n,Set(PAI=${CUT(PAI,@,1)})
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "anonymous"]?anonymous:cli)
exten => _+X.,n(anonymous),SIPAddHeader(P-Asserted-Identity: "${PAI}"
)
exten => _+X.,n,SIPAddHeader(Privacy: user\;id)
exten => _+X.,n,Goto(dial)
exten => _+X.,n(cli),SIPAddHeader(P-Asserted-Identity: "${PAI}"
)
exten => _+X.,n,SIPAddHeader(Privacy: id)
exten => _+X.,n,Goto(dial)
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[asterisk-users] 401 Unauthorized when originating SIP user exists on remote server
Hi list!
I'm trying to make a SIP test call from Bria and/or 3CXPhone from a PC
behind NAT.
From Bria/3CXPhone I connect to an Asterisk 11.25.0 server on the
internet at 100.100.94.210 with a SIP account "" created in sip.conf:
[]
type=friend
secret=something
host=dynamic
nat=yes
qualify=no
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=voipin
I dial +1234 which goes to remote-server:
exten => +1234,1,Dial(SIP/${EXTEN}@remote-server)
On remote-server (IP: 100.100.92.16), which is running Asterisk 10.7.1,
I have the following entry for 100.100.94.210 in sip.conf:
[incoming-server]
host=100.100.94.210
type=peer
insecure=port,invite
context=voipin
disallow=all
allow=alaw
canreinvite=no
dtmfmode=rfc2833
However, there's also a SIP account "" on that same server:
[]
type=friend
secret=something
host=dynamic
nat=yes
qualify=no
disallow=all
allow=alaw
allow=ulaw
canreinvite=no
context=voipin
When I make that call I get "401 Unauthorized" from remote-server. When
I modify the username on 100.100.94.210 to "" or anything else that
doesn't exist on 100.100.92.16, the call connects.
tcpdump:
100.100.94.210.5060 > 100.100.92.16.5060: SIP, length: 839
INVITE sip:+1234@100.100.92.16 SIP/2.0
Via: SIP/2.0/UDP 100.100.94.210:5060;branch=z9hG4bK0aa2c03b
Max-Forwards: 70
From: ;tag=as14990327
To:
Contact:
Call-ID: 6c33aa196f7a2c206a6b50a27b6a23d6@100.100.94.210:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.0
Shouldn't Asterisk check first for IP-based authentication and ignore
the From: part? In my case, use only the "incoming-server" peer.
Let's imagine remote-server would receive SIP calls which originate from
the PSTN... and the originating caller somewhere in the world uses
"" as username/CLI so that it makes it into the "From: sip:@"
part. That call would also get rejected with 401 Unauthorized if I'm not
mistaken?
Is there a switch I'm missing?
Thank you, as always!
Markus
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Re: [asterisk-users] cdr_mysql: Cannot connect to database server - SSL error: SSL_CTX_set_default_verify_paths failed
Hi, Am 08.06.2020 um 12:25 schrieb Antony Stone: On Monday 08 June 2020 at 12:15:56, Markus wrote: Hi list! I'm getting this error frequently: ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (2026) SSL connection error: SSL_CTX_set_default_verify_paths failed Hmmm... talking to localhost over SSL... seems a bit computationally wasteful. I switched to connecting via sock. My cdr_mysql.conf looks like this: [global] ;hostname=localhost dbname=asterisk table=cdr password=somepassword user=someuser ;port=3306 sock=/var/lib/mysql/mysql.sock However, I'm still frequently getting: "[Jun 29 12:40:08] ERROR[12977][C-07cc]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (2026) SSL connection error: SSL_CTX_set_default_verify_paths failed" Which makes no sense because I'm no longer connecting via localhost. The error message still mentions localhost, though. Maybe the error message is not accurate enough?! Strange? As explained previously, the error comes and go. Right now I'm losing about 10% of my CDRs. But that means it's not a general problem with the SSL stuff. Thanks! Markus -- _ -- 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] cdr_mysql: Cannot connect to database server - SSL error: SSL_CTX_set_default_verify_paths failed
Hi list! I'm getting this error frequently: ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to database server localhost: (2026) SSL connection error: SSL_CTX_set_default_verify_paths failed Right now, as a workaround, I reload Asterisk via cron once an hour, and after the reload everything is fine again _for a while_. Still, over the course of a month I lose about 12% of my CDRs this way. I guess I could reload more often but would prefer a real fix, not a workaround. Anyone got an idea? Asterisk 11.25.0 built by mockbuild @ olive.ph.tucny.com on a x86_64 running Linux on 2016-11-23 22:17:54 UTC Database is a Galera cluster: galera-3-25.3.19-2.el7.x86_64 Asterisk is writing to the local MySQL instance and there's SSL between the Galera cluster nodes. Thanks! Markus -- _ -- 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] 10 Caller IDs to be used randomly or progressively
Am 18.09.2019 um 01:15 schrieb bilal ghayyad:
I have 10 Caller IDs and I need each call (each time) to use one of
these Caller IDs to be the caller id.
exten => 1234,1,AGI(/var/lib/asterisk/randomcli.sh)
randomcli.sh:
#!/bin/bash
WORDFILE="/var/lib/asterisk/mobilecliall.txt"
NUMWORDS=1
#Number of lines in $WORDFILE
tL=`awk 'NF!=0 {++c} END {print c}' $WORDFILE`
for i in `seq $NUMWORDS`
do
rnum=$((RANDOM%$tL+1))
CIDNAME=$(sed -n "$rnum p" $WORDFILE)
done
echo $CIDNAME
echo "Setting CallerID to: \"\"<$CIDNAME>" >&2
echo "SET CALLERID \"\"<$CIDNAME>"
Then put your 10 CLIs in mobliecliall.txt.
PS: Shorter version for randomcli.sh, same functionality:
#!/bin/bash
CIDNAME=$(shuf -n 1 mobilecliall.txt)
echo "SET CALLERID \"\"<$CIDNAME>"
:-))
Regards
Markus
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Re: [asterisk-users] Playing a beep/noise during a call
I use ChanSpy for that.
This should get you on track:
https://community.asterisk.org/t/play-audio-file-on-channel-that-is-in-confbridge/67678
I don't use AMI, I just trigger asterisk binary through a shell script
(via AGI) like this to originate the call, it's easier for me:
/usr/sbin/asterisk -rx "originate local/something@whisper extension
something@whisper-playback"
[whisper]
exten => _[a-z,0-9].,1,Answer
exten => _[a-z,0-9].,n,Verbose(... whispering to admins)
exten => _[a-z,0-9].,n,ChanSpy(SIP/${EXTEN},qw)
exten => _[a-z,0-9].,n,Hangup
[whisper-playback]
exten => ,1,Answer
exten => ,n,Verbose(... playback to chanspy for admins)
exten => ,n,Playback(/var/lib/asterisk/sounds/somewavfile)
So, if you need this periodically you could maybe spawn a shell script
through AGI which will sleep for a certain amount of minutes and then
repeat the originate command ...
Good luck!
Am 07.02.2019 um 12:59 schrieb Rui Mota:
Hi.
I am using an appliance as SIP-ISDN gateway that has a (not so nice)
feature of hanging up a call if it detects more than 10 minutes of
silence from the originating side.
The problem is that the calls are always originated from a digital
system that, most of the times, just sends silence, for more than 10
minutes, only sporadically has people talking, so this makes the calls
to hangup automatically after 10 minutes...
I need to inject some noise/tone, periodically (like every 9 minutes),
on the originating call to avoid this behaviour. What is the best way to
achieve it?
I've seen the Extenspy or mixmonitor commands, but i need to make it
periodic...
Thank you in advance.
Rui
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[asterisk-users] ConfBridge: Identifying troublemakers
Hi list, imagine a ConfBridge conference with 10 participants. Now, one of them suddenly starts to yell and scream. Is there any built-in functionality (maybe not in ConfBridge, but in Asterisk itself) to identify the "loudest" caller? Or maybe already built-in functionality to automatically lower his speech volume in a case where he starts to scream, respectively adjust it to the level of the other participants? How about a scenario where that troublemaker does not scream, but instead starts to insult the other participants. He's not loud, just rude. I imagine a solution where a ConfBridge moderator/admin has to press a DTMF key and the ConfBridge will offer him a menu of that particular person who's talking at that very second. E.g... Caller: "You're all stupid! I don't like y..." - Moderator presses "5" in that very second ConfBridge talks only to Moderator: "Welcome to the moderator menu. The person who was talking in that very second is on channel SIP/something-123456 Please press 1 to kick him out. Please press 2 to ban him. Please press 3 to return to the conference." Something like that... Thanks for any pointers & tips! :) Regards Markus -- _ -- 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] Configure SIP reply timeout (timerb in sip.conf)
Reply to self: Found the problem after reading this post: http://lists.digium.com/pipermail/asterisk-dev/2010-March/042735.html You need to set timert1 in the peer config to *something*, otherwise it will ignore the timerb setting. Bug? It now looks like this and works fine: [peer01] host=1.2.3.4 type=peer context=nowhere disallow=all allow=alaw allow=ulaw canreinvite=no dtmfmode=rfc2833 timert1=500 timerb=2000 timert1=500 is the default anyway, according to sip.conf comments... Regards Markus Am 07.01.2019 um 17:23 schrieb Markus: Dear list, Asterisk 11.25.0 user here. I'm trying to set up failing over to a second SIP peer if the first SIP peer doesn't answer on our SIP INVITE within 2 seconds. In sip.conf I set timerb=2000 for this peer, but it doesn't seem to have any effect. The timeout is 6.5 seconds instead, which is in line with this description from sip.conf: "timerb: Call setup timer. If a provisional response is not received in this amount of time, the call will autocongest. Defaults to 64*timert1 (Which is 100 ms = rougly 6.5 seconds)" Maybe I cannot set timerb on a peer-basis? Here's my peer config: [peer01] host=1.2.3.4 type=peer context=nowhere disallow=all allow=alaw allow=ulaw canreinvite=no dtmfmode=rfc2833 timerb=2000 Thank you! Markus -- _ -- 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] Configure SIP reply timeout (timerb in sip.conf)
Dear list, Asterisk 11.25.0 user here. I'm trying to set up failing over to a second SIP peer if the first SIP peer doesn't answer on our SIP INVITE within 2 seconds. In sip.conf I set timerb=2000 for this peer, but it doesn't seem to have any effect. The timeout is 6.5 seconds instead, which is in line with this description from sip.conf: "timerb: Call setup timer. If a provisional response is not received in this amount of time, the call will autocongest. Defaults to 64*timert1 (Which is 100 ms = rougly 6.5 seconds)" Maybe I cannot set timerb on a peer-basis? Here's my peer config: [peer01] host=1.2.3.4 type=peer context=nowhere disallow=all allow=alaw allow=ulaw canreinvite=no dtmfmode=rfc2833 timerb=2000 Thank you! Markus -- _ -- 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] More testing - sorry guys
I received it :-) Am 28.03.2018 um 22:44 schrieb Matt Fredrickson: Just a test. -- _ -- 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] Half Off Topic Questions
Hi Group, we're just wondering, in German we call the different types of phone-numbers (Geographic,mobile,national,VoIP...) Rufnummerngassen (phone number alleys ;-) ) Is there an english word for this? -- - Markus Weiler markus_wei...@mailworks.org - -- _ -- 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] res_json
Hi All, this seems to be a really neat module, that could really help us. https://github.com/drivefast/asterisk-res_json Any opinions about if we should use it in a production system? Maybe from "official" asterisk side? thanks!! Markus -- _ -- 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] Call preemption
The task itself sounds like a job for an AGI script to me... check for amount of calls, if 10, hangup one. But how do you determine the priority of a call? Am 07.11.2017 um 12:21 schrieb Jean Aunis: Hello, Has anyone already implemented some sort of call preemption in Asterisk ? I am trying to achieve something like this : [...] Does anyone have an idea ? -- _ -- 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] Blocking outgping caller id on a PRI E1
Can you try exten => s,n,Set(CALLERID(num)=anonymous) in your dialplan before passing the call to the PRI? and/or exten => s,n,Set(CALLERID(name)=anonymous) and/or exten => s,n,Set(CALLERID(all)=anonymous) If it doesn't work, maybe replace the "anonymous" with "" or something else. Sorry, no guarantee, just a shot in the dark. Am 08.11.2017 um 16:02 schrieb Neil Youngman: I am trying to block/hide outgoing caller id on a PRI E1. It seems that it should be fairly simple, but it is defeating me. https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID says: "to hide your caller id, use: Set(CALLERID(num-pres)=prohib)" That doesn't seem to do it. The calls are originated from AMI and I have tried a blank "CallerId:" line and "CallerId: <>"". Neither of those made any difference. I have also tried "hidecallerid=yes" in chan_dahdi.conf, but that has also made no difference. I assume that I am missing something obvious? Neil Youngman Neil Youngman Developer Wirefast Limited Wirefast provides secure corporate messaging services. See our messaging solutions at http://www.wirefast.com/ Please consider the environment. Does this email or attachment need to be printed? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Any views or opinions are solely those of the author and do not necessarily represent those of Wirefast Limited Email transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. Wirefast Limited is registered in England & Wales Company number: 03865860 Registered Office: 7/10 Chandos Street, Cavendish Square, London, W1G 9DQ Wirefast definitions of classification can be found here: www.wirefast.com/classifications -- _ -- 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] Turn on SIP debugging from DialPlan
While we're at it, check out sngrep. Alex B. mentioned it on another mailing list a couple days ago. Screenshots: https://github.com/irontec/sngrep/wiki/Screenshots Download: https://github.com/irontec/sngrep Am 18.02.2017 um 05:10 schrieb Markus Weiler: Hi Derek, I think Homer (http://sipcapture.org/) is the right answer :-) HEP Agent will send the SIP trace to a remote Server (res_hep). Markus Am 18.02.2017 um 00:18 schrieb Tim Pozar: You can tell it to just capture SIP traffic and not the RTP traffic. Nice write up of using TCPdump and wireshark can be found here: https://blog.flowroute.com/2014/04/10/how-to-capture-sip-packets/ BTW, I have found this works really well in trying to debug RTP traffic as well. Wireshark just does the right thing in putting audio back together. Very helpful in tracking down in and out of band DTMF problems that we were having with various carriers. Tim On 2/17/17 3:07 PM, Derek Andrew wrote: The SIP trace will be adequate but this is on a remote system with limited disk space. I would love to turn on debugging while making the troublesome calls, then turn it off afterward. Tcpdump is great, but starting it and stopping it and keeping all that data would still be an issue. d On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <po...@lns.com <mailto:po...@lns.com>> wrote: Why not capture the packets with something like tcpdump and run it through Wireshark? Tim On 2/17/17 2:43 PM, Derek Andrew wrote: > I have some troublesome numbers that I would like to capture the SIP > dialogue when I am calling them. When I am about to dial the number, is > there any way to turn on SIP debugging in the dial plan before I make > the call? (and turn it off after the call is completed?) > > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <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 <http://lists.digium.com/mailman/listinfo/asterisk-users> -- Copyright 2017 Derek Andrew (excluding quotations) +1 306 966 4808 Communication and Network Services Information and Communications Technology Infrastructure Services *University of Saskatchewan *Peterson 120; 54 Innovation Boulevard Saskatoon,Saskatchewan,Canada. S7N 2V3 Timezone GMT-6 Typed but not read. -- _ -- 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] Turn on SIP debugging from DialPlan
Hi Derek, I think Homer (http://sipcapture.org/) is the right answer :-) HEP Agent will send the SIP trace to a remote Server (res_hep). Markus Am 18.02.2017 um 00:18 schrieb Tim Pozar: You can tell it to just capture SIP traffic and not the RTP traffic. Nice write up of using TCPdump and wireshark can be found here: https://blog.flowroute.com/2014/04/10/how-to-capture-sip-packets/ BTW, I have found this works really well in trying to debug RTP traffic as well. Wireshark just does the right thing in putting audio back together. Very helpful in tracking down in and out of band DTMF problems that we were having with various carriers. Tim On 2/17/17 3:07 PM, Derek Andrew wrote: The SIP trace will be adequate but this is on a remote system with limited disk space. I would love to turn on debugging while making the troublesome calls, then turn it off afterward. Tcpdump is great, but starting it and stopping it and keeping all that data would still be an issue. d On Fri, Feb 17, 2017 at 4:56 PM, Tim Pozar <po...@lns.com <mailto:po...@lns.com>> wrote: Why not capture the packets with something like tcpdump and run it through Wireshark? Tim On 2/17/17 2:43 PM, Derek Andrew wrote: > I have some troublesome numbers that I would like to capture the SIP > dialogue when I am calling them. When I am about to dial the number, is > there any way to turn on SIP debugging in the dial plan before I make > the call? (and turn it off after the call is completed?) > > > > > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ <https://community.asterisk.org/> New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started <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 <http://lists.digium.com/mailman/listinfo/asterisk-users> -- Copyright 2017 Derek Andrew (excluding quotations) +1 306 966 4808 Communication and Network Services Information and Communications Technology Infrastructure Services *University of Saskatchewan *Peterson 120; 54 Innovation Boulevard Saskatoon,Saskatchewan,Canada. S7N 2V3 Timezone GMT-6 Typed but not read. -- _ -- 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] Issue with handling of 480 DND
Am 06.01.2017 um 12:07 schrieb Markus Weiler:
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
We tried to figure out what the difference is and think it's how
Asterisk handles the "480 Do Not Disturb" from the phone
(xxx.xxx.xxx.xxx).
Is this a bug or a feature? :-) How could we handle this correctly?
I *think* you can parse HANGUPCAUSE to catch that? But really not sure.
Here's a snippet I stole from somewhere to do failover to a 2nd gateway:
exten =>
_something.,n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"|"${DIALSTATUS}"="CONGESTION"|"${HANGUPCAUSE}"="34"]?failover)
Not really sure how or if that works, but googling Asterisk Hangupcause
should get you started?
Good luck.
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Re: [asterisk-users] Issue with handling of 480 DND
Nobody any idea?
It would be really helpful,
Markus
Am 06.01.2017 um 12:07 schrieb Markus Weiler:
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
Stripped down example;
exten = 494X,n,Dial(SIP/4120089,15,w)
exten = 494X,n,Goto(98-${DIALSTATUS},1)
exten = 494X,n,Hangup()
.
exten = 98-BUSY,1,NoOp(Busy)
exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))
exten = 98-NOANSWER,1,NoOp(noanswer)
exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))
Íf the phone call times out, the call is sent to 98-NOANSWER and then
answered as expected.
If the User presses DND on his phone the call is sent to 98-BUSY which
is identical but then the call is hung up. This behaviour is
unexpected/unwanted.
We tried to figure out what the difference is and think it's how
Asterisk handles the "480 Do Not Disturb" from the phone
(xxx.xxx.xxx.xxx).
It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
is being forwarded".
Is this a bug or a feature? :-) How could we handle this correctly?
SIP and Asterisk debug log below. Any help would be appreciated!
Markus
SIP:
#
U 2017/01/06 11:38:29.515836 xxx.xxx.xxx.xxx:45731 ->
yyy.yyy.yyy.yy:5060
SIP/2.0 480 Do Not Disturb.
v: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport=5060.
f: "0160XXX" <sip:0160...@yyy.yyy.yyy.yy>;tag=as4ef364e1.
t: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
i: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 INVITE.
User-Agent: snom760/8.7.5.13.
m: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;reg-id=1.
l: 0.
.
#
U 2017/01/06 11:38:29.516045 yyy.yyy.yyy.yy:5060 ->
xxx.xxx.xxx.xxx:45731
ACK sip:4120089@192.168.178.70:45731;line=8lln9qsq SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport.
Max-Forwards: 70.
From: "0160XXX" <sip:0160...@yyy.yyy.yyy.yy>;tag=as4ef364e1.
To: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
Contact: <sip:0160...@yyy.yyy.yyy.yy:5060>.
Call-ID: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 13.12.1.
Content-Length: 0.
.
#
U 2017/01/06 11:38:29.516166 yyy.yyy.yyy.yy:5060 -> zzz.zzz.zzz.zzz:5060
SIP/2.0 181 Call is being forwarded.
Via: SIP/2.0/UDP
zzz.zzz.zzz.zzz;branch=z9hG4bK120a.dcdd311bf80a9536d4eec1ce380f66a7.0;received=zzz.zzz.zzz.zzz.
Via: SIP/2.0/UDP 93.189.169.102:5060;branch=z9hG4bK4ec14865.
Record-Route: <sip:zzz.zzz.zzz.zzz;lr=on;ftag=as47cd4bd4>.
From: "49160XXX" <sip:49160XXX@93.189.169.102>;tag=as47cd4bd4.
To: <sip:494xx...@zzz.zzz.zzz.zzz>;tag=as0823beee.
Call-ID: 15f8ee0e657ba8687db29cff0093c95b@93.189.169.102:5060.
CSeq: 102 INVITE.
Server: Asterisk PBX 13.12.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:494xx...@yyy.yyy.yyy.yy:5060>.
Content-Length: 0.
.
Asterisk Debug:
[Jan 6 11:38:29] VERBOSE[5383][C-000473c5] pbx.c: Executing
[494X@4120:7] Dial("SIP/SER_IB-0004cd6d", "SIP/4120089,15,w") in
new stack
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Asked to create a
SIP channel with formats: (alaw)
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Allocating new SIP
dialog for 0dac947057f4ddc613b1391948770...@yyy.yyy.yyy.yy:5060 - INVITE
(No RTP)
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: Using engine
'asterisk' for RTP instance '0x7fae14073678'
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Allocated
port 14098 for RTP instance '0x7fae14073678'
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: RTP instance
'0x7fae14073678' is setup and ready to go
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] acl.c: Not an IPv4 nor IPv6
address, cannot get port.
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: Splitting 'pbx6'
into...
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: ...host 'pbx6' and
port ''.
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Setup RTCP
on RTP instance '0x7fae14073678'
[Jan 6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
TOS bits 184
[Jan 6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
CoS mark 5
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] acl.c: For destination
'xxx.xxx.xxx.xxx', our source address is 'yyy.yyy.yyy.yy'.
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting
AST_TRANSPORT_UDP with address yyy.yyy.yyy.yy:5060
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan 6 11:38:29] DEBUG[5383][C-0004
[asterisk-users] Issue with handling of 480 DND
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
Stripped down example;
exten = 494X,n,Dial(SIP/4120089,15,w)
exten = 494X,n,Goto(98-${DIALSTATUS},1)
exten = 494X,n,Hangup()
.
exten = 98-BUSY,1,NoOp(Busy)
exten = 98-BUSY,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))
exten = 98-NOANSWER,1,NoOp(noanswer)
exten = 98-NOANSWER,n,ExecIf($["${Voicemail}" =
"1"]?Playback(/home/4120/mitarbeiter/ab))
Íf the phone call times out, the call is sent to 98-NOANSWER and then
answered as expected.
If the User presses DND on his phone the call is sent to 98-BUSY which
is identical but then the call is hung up. This behaviour is
unexpected/unwanted.
We tried to figure out what the difference is and think it's how
Asterisk handles the "480 Do Not Disturb" from the phone
(xxx.xxx.xxx.xxx).
It is passed to our main incoming server (zzz.zzz.zzz.zzz) as "181 call
is being forwarded".
Is this a bug or a feature? :-) How could we handle this correctly?
SIP and Asterisk debug log below. Any help would be appreciated!
Markus
SIP:
#
U 2017/01/06 11:38:29.515836 xxx.xxx.xxx.xxx:45731 ->
yyy.yyy.yyy.yy:5060
SIP/2.0 480 Do Not Disturb.
v: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport=5060.
f: "0160XXX" <sip:0160...@yyy.yyy.yyy.yy>;tag=as4ef364e1.
t: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
i: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 INVITE.
User-Agent: snom760/8.7.5.13.
m: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;reg-id=1.
l: 0.
.
#
U 2017/01/06 11:38:29.516045 yyy.yyy.yyy.yy:5060 ->
xxx.xxx.xxx.xxx:45731
ACK sip:4120089@192.168.178.70:45731;line=8lln9qsq SIP/2.0.
Via: SIP/2.0/UDP yyy.yyy.yyy.yy:5060;branch=z9hG4bK749dbc68;rport.
Max-Forwards: 70.
From: "0160XXX" <sip:0160...@yyy.yyy.yyy.yy>;tag=as4ef364e1.
To: <sip:4120089@192.168.178.70:45731;line=8lln9qsq>;tag=0380h4r478.
Contact: <sip:0160...@yyy.yyy.yyy.yy:5060>.
Call-ID: 7568eb9e7c148e535166a89702423...@yyy.yyy.yyy.yy:5060.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 13.12.1.
Content-Length: 0.
.
#
U 2017/01/06 11:38:29.516166 yyy.yyy.yyy.yy:5060 -> zzz.zzz.zzz.zzz:5060
SIP/2.0 181 Call is being forwarded.
Via: SIP/2.0/UDP
zzz.zzz.zzz.zzz;branch=z9hG4bK120a.dcdd311bf80a9536d4eec1ce380f66a7.0;received=zzz.zzz.zzz.zzz.
Via: SIP/2.0/UDP 93.189.169.102:5060;branch=z9hG4bK4ec14865.
Record-Route: <sip:zzz.zzz.zzz.zzz;lr=on;ftag=as47cd4bd4>.
From: "49160XXX" <sip:49160XXX@93.189.169.102>;tag=as47cd4bd4.
To: <sip:494xx...@zzz.zzz.zzz.zzz>;tag=as0823beee.
Call-ID: 15f8ee0e657ba8687db29cff0093c95b@93.189.169.102:5060.
CSeq: 102 INVITE.
Server: Asterisk PBX 13.12.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Session-Expires: 1800;refresher=uas.
Contact: <sip:494xx...@yyy.yyy.yyy.yy:5060>.
Content-Length: 0.
.
Asterisk Debug:
[Jan 6 11:38:29] VERBOSE[5383][C-000473c5] pbx.c: Executing
[494X@4120:7] Dial("SIP/SER_IB-0004cd6d", "SIP/4120089,15,w") in
new stack
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Asked to create a
SIP channel with formats: (alaw)
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Allocating new SIP
dialog for 0dac947057f4ddc613b1391948770...@yyy.yyy.yyy.yy:5060 - INVITE
(No RTP)
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: Using engine
'asterisk' for RTP instance '0x7fae14073678'
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Allocated
port 14098 for RTP instance '0x7fae14073678'
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] rtp_engine.c: RTP instance
'0x7fae14073678' is setup and ready to go
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] acl.c: Not an IPv4 nor IPv6
address, cannot get port.
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: Splitting 'pbx6'
into...
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] netsock2.c: ...host 'pbx6' and
port ''.
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] res_rtp_asterisk.c: Setup RTCP
on RTP instance '0x7fae14073678'
[Jan 6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
TOS bits 184
[Jan 6 11:38:29] VERBOSE[5383][C-000473c5] netsock2.c: Using SIP RTP
CoS mark 5
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] acl.c: For destination
'xxx.xxx.xxx.xxx', our source address is 'yyy.yyy.yyy.yy'.
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting
AST_TRANSPORT_UDP with address yyy.yyy.yyy.yy:5060
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: Setting NAT on RTP
to On
[Jan 6 11:38:29] DEBUG[5383][C-000473c5] chan_sip.c: SIP call-id
changed from '0dac947057f4ddc613b1391948770...@yyy.yyy.yyy.yy:5060' to
'756
Re: [asterisk-users] Asterisk dahai install centos 7
There are inofficial RPMs for CentOS 7 available if you don't want to mess with compiling: https://www.tucny.com/telephony/asterisk-rpms Am 10.12.2016 um 15:47 schrieb christopher kamutumwa: Hello support am trying to install dahdi on centos 7 and am doing the make ommand and below is result any way out this You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel installed. uname -a gives me below; Linux localhost.localdomain 3.10.0-327.el7.x86_64 #1 SMP Thu Nov 19 22:10:57 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux [root@localhost dahdi-linux-complete-2.11.1+2.11.1]# make make -C linux all make[1]: Entering directory `/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 3.10.0-327.el7.x86_64 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux' make: *** [all] Error 2 -- _ -- 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] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)
Am 28.10.2016 um 17:58 schrieb Max Grobecker:
why not using FILTER() in your dialplan to eleminate all chars that are not
numeric?
Like
Set(VAR=${FILTER(0-9+),${EXTEN}})
That would eleminate all characters you're not expecting.
That's great! Didn't know FILTER. Thanks!
--
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/
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[asterisk-users] Just got defrauded - how do I block calls which contain a dash (RegEx noob question)
Hi list, I'm using Asterisk2Billing (v2.0.16) and it appears to have an annoying bug. When there are rates for e.g. 44 (UK landline) and 44870 (UK premium) and a fraudster manages to somehow dial 44-870 instead of 44870 the rate for 44 will match, not the one for 44870. So, I would like to block all calls on a dialplan level that contain a dash. -44, 4-4, 44-, 44---, -, ---, just everything with a friggin' dash. My noob-ish try: exten => _-.,1,NoOp(Blocking dash) exten => _-.,n,Hangup Doesn't work. On https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching I found: "The dash (-) character is ignored in extensions and patterns except when it is used in a pattern to specify a range in a character set. It has no effect in matching or sorting extensions." How do I do it right? Thank you! Markus -- _ -- 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] Including doesn't have any effect
Hi AJ, Am 06.06.2016 um 10:14 schrieb A J Stiles: But why not call an AGI script, have this check the caller ID against a MySQL database and return a status -- blocked or not -- in a variable? Then you can manage individual number blocking in a much cleaner, more extensible fashion. Feel free to ignore me if it sounds like I'm suggesting you walk all the way to the tool shed to fetch a chisel, when you know the screwdriver in your drawer is already up to the job :) you're right, it would be the better solution! But I'm simply too lazy to implement that. :D Regards Markus -- _ -- 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] Including doesn't have any effect
Am 04.06.2016 um 21:58 schrieb Steve Edwards: 1) If the caller ID matches '+493456789' (the first one), you goto the 'hangup' label. If it does not match, you goto the 'nohangup' label -- skipping the subsequent tests. Doh! Removed :nohangup from every line but the last one and now it works as it should. Thank you! :) -- _ -- 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] Including doesn't have any effect
Hi list,
n00b question, but I can't figure it out:
[callthrough]
exten => _+X.,1,NoOp(nothing here)
#include "blockedall.conf"
exten => _+X.,n(hangup),Hangup
exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" =
"anonymous"]?nocli:cli)
... more stuff that is handling the call ...
I'm putting CLIs that I don't want to be able to call my system into
blockedall.conf:
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "+493456789"]?hangup:nohangup)
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "+492345678"]?hangup:nohangup)
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "+491234567"]?hangup:nohangup)
But it never moves to "hangup" when I call from any of those CLIs :-(
BUT, if I include it directly in extensions.conf, it works:
[callthrough]
exten => _+X.,1,NoOp(nothing here)
exten => _+X.,n,GotoIf($["${CALLERID(num)}" = "+493456789"]?hangup:nohangup)
exten => _+X.,n(hangup),Hangup
exten => _+X.,n(nohangup),GotoIf($["${CALLERID(num)}" =
"anonymous"]?nocli:cli)
... more stuff that is handling the call ...
I made sure that "blockedall.conf" actually gets included by executing
"dialplan show callthrough".
What am I missing?
Asterisk 11.17.1.
Thank you!
Markus
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Re: [asterisk-users] asterisk server stress test
Am 20.08.2015 um 03:16 schrieb Pete Mundy: Ah cr@p, sorry Steve, didn't mean to top-post there. On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org mailto:markus_wei...@mailworks.org wrote: We started the 500 calls and used milliwatt app on the first and record on the second host to check the quality. Alternatively just start 500+ calls and call yourself on top. So you can get a good idea how the quality is. Markus That's a fascinating concept! Can you share any more about how you appraised the data and determined your results? ie once you had the recordings on the second host what did you do do computationally score them? Do you look at the decoded (1khz?) waveform or do you appraise in another way? Pete Hi Pete, we used different approaches. Just to test the maximum channels a gateway can process the two Methods are enough, you can either listen to the Recordings or look at the waveform. The easiest approach is to call a colleague and gradually increase the calls on the machine. For systematic, continuous analysis Voipmonitor is a very useful tool. We directed the traffic to a mirroring port on the Switch to which we connected a Server running Voipmon. (http://www.voipmonitor.org/) Voipmon records the call and rates its quality. You can check the results either using the commercial Web Interface (test for free) or query the mysql DB. Unfortunately Voipmon tends to crash on a regular basis (at least when we used it), but it's an awesome tool. The underlying tool pcapsipdump is running a lot more stable, but you need to put a lot more work into it to get started. hope i could help Markus -- _ -- 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 server stress test
Am 19.08.2015 um 19:07 schrieb Steve Edwards:
Please don't top post.
On Wed, 19 Aug 2015, James Cass wrote:
Steve, would you be willing to share that quick bash script?
There's no magic in the script, but here it is, embarrassing myself:
cp sample-call-file /tmp/
chmod +x /tmp/sample-call-file
for I in $(seq 1 $1)
do
sudo -u asterisk\
cp /tmp/sample-call-file\
/var/spool/asterisk/outgoing/${RANDOM}
done
sleep 10
Here's what's wrong with this snippet:
1) I don't know why I chmod the 'template.' No idea whatsoever.
Alcohol may have been involved.
2) I hate single character variable names. I love alcohol.
3) cp is ill advised. For a testing script, it was easy. For a
production application, use mv.
In use, I would execute it specifying how many call files to create,
like 50. Then, take a look at top, iftop, and vmstat. Lather, rinse,
repeat to get to your goal.
We started the 500 calls and used milliwatt app on the first and record
on the second host to check the quality. Alternatively just start 500+
calls and call yourself on top. So you can get a good idea how the
quality is.
Call-Files are explained on
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Markus
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Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining
Hi Stefan, we ran into a similar problem using Debian. There we are able to check the current limits using: pidof asterisk - 23351 cat /proc/23351/limits Output: Limit Soft Limit Hard Limit Units Max open files1024 1024 files I think that in the end /etc/security/limits.conf * hardnofile 50 * softnofile 50 root hardnofile 50 root softnofile 50 did the trick. We also tried vi /etc/sysctl.conf fs.file-max = 50 not sure what the solution in the end was. But I remember rebooting was important. Markus Am 11.08.2015 um 11:00 schrieb Stefan Viljoen: Anybody else ran into this? No, but I would ask myself why so many file descriptors are being used. It sounds like you have a file descriptor leak (not being closed when finished with). Hi Tony Thanks for replying. I suspected something like that, though repeatedly running lsof | wc -l Always stays quite low - 100 000 open files, which is still 8 times less than the system maximum as confirmed by running ulimit -n I also note that this number will increase to about 125 000 but never go higher than that, then, as calls hang up, decreate again - during times when the CLI is spammed with 100s of broken pipe errors due to insuffiecient file descriptors, this number never reaches beyond 125 000 out of the available 800 000 open files. You might also want to look at the output of lsof (or at least some of it) to see what all these file descriptors are pointing to, and whether it is indeed Asterisk that is consuming them. If I grep by asterisk on the output of lsof the few thousand lines I have looked at all seem to indicate legitimate uses - there are at least two files for each conversation in progress (I assume for inward and outward RTP) plus one for each file being mixmonitored (which also seems logical) and also number-of-active-calls connections to res_timing_dahdi - which all looks correct... If it is Asterisk, it's quite possible, even probable, that such a leak has been found and fixed, even in the 1.8 series. 1.8.11.0 is rather old - the latest is 1.8.32.3, so it would be best to update to that version and see if the problem persists. Ok, I will have to consider that. The thing is the problem is not consistent - I can (for example) run 60 calls, with no problems and no reported failures in opening files, then calls will -decrease- to about 40 and then later spike to 70, but around 50 calls I get the errors coming up thousands of times in the CLI, then suddenly stop as the calls -increase- which doesn't make sense. But this kind of behaviour does seem consistent with a possible leak. SOMETHING NEW I have now ran /usr/bin/prlimit --pid `pidof asterisk` and I have noticed that even though I have 800 000 files specified, the ACTUAL limit in place on Asterisk for numbers of files is only 1024?! # prlimit --pid `pidof asterisk` RESOURCE DESCRIPTION SOFT HARD UNITS AS address space limitunlimited unlimited bytes CORE max core file size unlimited unlimited blocks CPUCPU time unlimited unlimited seconds DATA max data size unlimited unlimited bytes FSIZE max file size unlimited unlimited blocks LOCKS max number of file locks held unlimited unlimited MEMLOCKmax locked-in-memory address space 65536 65536 bytes MSGQUEUE max bytes in POSIX mqueues819200819200 bytes NICE max nice prio allowed to raise 0 0 NOFILE max number of open files1024 4096 NPROC max number of processes30861 30861 RSSmax resident set size unlimited unlimited pages RTPRIO max real-time priority 0 0 RTTIME timeout for real-time tasksunlimited unlimited microsecs SIGPENDING max number of pending signals 30861 30861 STACK max stack size 8388608 unlimited bytes Accordingly I have put this into a cronjob ran each minute: prlimit --pid `pidof asterisk` --nofile=786000:786000 to try and force the running binary to keep a high file limit (sources say to keep it less than the ACTUAL system file limit, in my case 800 000 files) on the live Asterisk process. I'll see if this maybe helps - the above runs via cron each minute. So it appears for some reason somehow the live running asterisk process loses track of how many open files it may have, or when it starts it somehow does not start with the correct number of maximum open files, as set in the system / kernel config? Anyway, thank you for replying, I'll monitor this new Cronjob fixup I'm trying and see if it helps. No wonder it is complaining about running out of file handles if it ACTUALLY was only
Re: [asterisk-users] asterisk email to fax
Hi Eric,
This is not really easy. Especially the Mail2Tiff Conversion is tricky
(lots of different MIME/File formats).
When you have the correct tiff file the rest ist easy. Try to narrow it
down to an empty Mail Body using (one) PDF attachment.
We used a self written Java app to prepare the files.
Markus
Am 25.06.2015 um 18:09 schrieb Eric Cooper:
On Wed, Jun 24, 2015 at 08:15:06AM +0200, tux john wrote:
i would like to add email to fax functionality to the system. could someone
point me to the right direction to see how please?
I don't have a general solution, since I haven't needed to send faxes
recently. But I did set up my dialplan so that I could test that it
works.
I used this:
;; For outgoing faxes
FAX_NAME = My Company Name
FAX_NUMBER = My Phone Number
FAX_FILE = /tmp/fax.tiff
[services]
...
exten = sendfax,1,Verbose(3,Sending fax)
same = n,Set(FAXOPT(headerinfo)=${FAX_NAME})
same = n,Set(FAXOPT(localstationid)=${FAX_NUMBER})
same = n,SendFax(${FAX_FILE})
same = n,Verbose(3,Pages: ${FAXOPT(pages)} Status: ${FAXOPT(status)})
same = n,System(rm ${FAX_FILE})
same = n,Hangup()
To dial the sendfax extension, I created a call file manually that
looked like this:
Channel: DAHDI/g0/1-800-123-4567
Context: services
Extension: sendfax
In an automated system, you'd probably want to use the manager API,
but you could also generated call files from scripts. Then I placed
the fax file in /tmp/fax.tiff and the call file in
/var/spool/asterisk/outgoing/ and Asterisk sent off the fax. There
are some free fax echo numbers out there that are useful when
debugging.
If I wanted to drive this automatically from email, I would probably
just use procmail and some scripts. Ghostscript produces the right
kind of image file if you use -sDEVICE=tiffg4. You'd have to define
some syntax for specifying the outgoing number and cover page info in
the email body, and extract the MIME attachment for the document from
the email.
Hope this helps.
--
Eric Cooper e c c @ c m u . e d u
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Re: [asterisk-users] German sounds on Asterisk
great, would be the ideal time to comment the www.voip-info.org to contribute to the community :-) Markus Am 14.06.2015 um 09:42 schrieb Luca Bertoncello: Markus Weiler markus_wei...@mailworks.org schrieb: Hi from voipinfo... If an Asterisk command specifies a sound file in a*subdirectory*, Asterisk looks in that subdirectory for the language subdirectory. For example, theSayDigits http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigitscommand may play the sound file digits/6. Asterisk will, if the language code is de, first look for /var/lib/asterisk/sounds/*digits/de/*6.gsm before falling back to /var/lib/asterisk/sounds/digits/6.gsm. Of course I read this page, but it does NOT work so... Right now I got it, using a new structure: /var/lib/asterisk/sounds/de/ /var/lib/asterisk/sounds/de/digits /var/lib/asterisk/sounds/de/letters /var/lib/asterisk/sounds/de/phonetics and it works... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] German sounds on Asterisk
Hi,
from voipinfo...
If an Asterisk command specifies a sound file in a*subdirectory*,
Asterisk looks in that subdirectory for the language subdirectory. For
example, theSayDigits
http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigitscommand may
play the sound file digits/6. Asterisk will, if the language code is
de, first look for /var/lib/asterisk/sounds/*digits/de/*6.gsm before
falling back to /var/lib/asterisk/sounds/digits/6.gsm.
Markus
http://www.voip-info.org/wiki/view/Asterisk+multi-language
Am 14.06.2015 um 09:36 schrieb Luca Bertoncello:
Hi again
I'd like to configured my Asterisk to use german sounds for the
Say-commands...
I installed the sounds-files and I tried them with
Playback(de/demo-echodone) and it works.
Now I tried to add an extension to say the current time:
exten = 24,1,Verbose(2,Time asked by ${CALLERID(num)})
Exten = 24,n,Set(CHANNEL(language)=de)
Exten = 24,n,SayUnixTime()
Exten = 24,n,Hangup
But if I call the 24, it says the time in English...
On the CLI I see:
-- Executing [24@default:2] Set(SIP/004935-0003,
CHANNEL(language)=de) in new stack
-- Executing [24@default:3] SayUnixTime(SIP/004935-0003, )
in new stack
-- SIP/004935-0003 Playing 'digits/day-0.gsm' (language 'de')
-- SIP/004935-0003 Playing 'digits/h-14.gsm' (language 'de')
...
So, it seems, it would use the German sounds, but it doesn't...
Has someone an explanation why it works so?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
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Re: [asterisk-users] Asterisk and Deutsche Telekom
Am 13.06.2015 um 14:44 schrieb Luca Bertoncello: If my Asterisk-configuration don't work, I don't have a phone and my wife cannot work... You will find out the day the switchover will be made by Telekom. If it doesn't work, you'll analyze why and fix it within a few minutes, you already did most of the preparation/work today, and then you're all set. I don't see the problem. It's not like you are running an emergency room or something where you depend on 24x7 connectivity. And if you were, you would approach this totally differently. -- _ -- 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 and Deutsche Telekom
Am 13.06.2015 um 13:54 schrieb Luca Bertoncello: I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). I don't think so. Most users will use the router provided by Telekom. Anyway, after 15 seconds of Google'ing for Magenta Zuhause and SIP, maybe this will help you: http://wiki.innovaphone.com/index.php?title=Howto:Deutsche_Telekom_Magenta_Zuhause_SIP_Provider_Compatibility_Test http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/faq-350884716;jsessionid=0071C998DA2C6AF06C6A4531D3EC952A If you can't test it right now, obviously, then just test it when the switchover has been made. What's the problem? :) -- _ -- 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] Strange and complete failure of Asterisk 1.8
definitely DNS... check your Register lines... Markus Am 27.05.2015 um 20:14 schrieb Duncan Turnbull: DNS failure could do this Asterisk used to get stuck in a symmetric DNS request wait state which meant everything ground to a halt as it waited for a reply while DNS timed out. The recommended option was either ip only or a DNS proxy that failed fast this letting asterisk continue Cheers Duncan On 27/05/2015, at 11:55 pm, Stefan Viljoen viljo...@verishare.co.za wrote: Hi all We've had a very strange failure on an Asterisk 1.8 install that has been running for about a year at a customer site. The physical hardware is fine, all other services off the Centos 6.5 server are running. Only Asterisk is not working... The first symptom was that no calls can be made over the SIP phones used with it, and no calls could be received over the SIP trunk connected to it. I checked and noted that sip show peers in the CLI would either do nothing (e. g. just show asterisk*cli again, with no response) or it would return only this: asterisk*CLI sip show peers Name/username HostDyn Forcerport ACL Port Status asterisk*CLI A module show like sip also does literally nothing, just asterisk*cli module show like sip asterisk*CLI Soon after this, I lost the capacity to get any response if I do an asterisk -r on the commandline - it would just hang indefintely. Did a reboot, and then, I couldnt start asterisk at all - entering # asterisk would also just hang. So, I recompiled asterisk from source and reinstalled the executable and all the module files. Still the same. I happened to have an older asterisk executable from a few months before laying around and sha256summed it - and there was a difference in the checksum vs. the non-working asterisk binary - BUT it turned out that the newly recompiled asterisk binary has the SAME SHA256 checksum as the non-working asterisk binary. System seems fine otherwise, nothing relevant in /var/log/messages or dmesg indicating a hardware failure. /var/log/asterisk/messages also contains no strange warnings or errors. Anybody got any idea why I cannot resuscitate my Asterisk install, even after recompiling it from scratch from source? Why would asterisk die like this in the first place? 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 -- _ -- 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
[asterisk-users] Asterisk API
Hi all, currently we're looking to program a new asterisk application. Years ago we used AMI and Asterisk Java. When we did this we pretty soon encountered performance issues when using a lot of channels. We want to place calls, bridge channels, disconnect channels, monitor them, hangup. What's the status with Asterisk REST API? Any experiences on performance,stability,documentation, caveats? Any toolkits for a fast start, Frameworks in any language? Hints? Best practices? Thanks for any insights! Markus -- _ -- 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] 603 Declined Dialstatus Busy
Hi Nick,
maybe this will help?
exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code :
${HASH(SIP_CAUSE,${CDR(dstchannel)})})
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause)
Markus
Am 27.02.2015 um 18:56 schrieb Nick Olsen:
Hello Everyone.
In my outbound contexts, I'm using ${DIALSTATUS} to fail over to
other routes if the chosen route rejects the call.
Now, My current scenario is if I get BUSY back from the first
provider, I send a busy back to my customer. If I get something like
CHANUNAVAIL (Like a SIP 503) I advance to the next carrier and attempt
the call.
This works great as expected. However, One of my SIP carriers likes to
send back 603 DECLINED inplace of 503's. Asterisk ${DIALSTATUS}
treats this as Busy. Can I change how asterisk interprets a 603
Declined? So it treats is as CHANUNAVAIL?
The obvious quick fix is to change my Busy option to attempt another
carrier before finally returning BUSY to the customer. But I was
hoping to not have to do that. Any ideas?
Nick Olsen
Network Operations
(855) FLSPEED x106
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Re: [asterisk-users] How to route SIP provider without DID
Am 17.02.2015 um 23:44 schrieb hbk:
I'm struggling to separate inbound calls fro a SIP provider that does
not send DID.
I have tried ...sip.com/12345678 on register string
different context=from-no-did
Port not possible as only support 5060
You're right, this is always an annoying and confusing scenario. Here's
my sample for sipgate which works for separating inbound and outbound:
sip.conf:
register = user:p...@sipgate.de/sipgate-in
[sipgate-out]
type=friend
insecure=invite
nat=no
username=user
fromuser=user
fromdomain=sipgate.de
secret=pass
host=sipgate.de
qualify=no
canreinvite=no
dtmfmode=rfc2833
context=sipgate
extensions.conf:
[sipgate]
exten = sipgate-in,1,NoOp
exten = sipgate-in,n,Dial(SIP/priv)
(This is for incoming calls only)
And for my SIP hardphone which receives the calls from sipgate and dials
out via sipgate:
sip.conf:
[priv]
type=friend
secret=anotherpass
host=dynamic
nat=no
disallow=all
allow=ulaw
allow=alaw
canreinvite=no
context=sipgate-priv
[sipgate-priv]
exten = _X.,1,NoOp
exten = _X.,n,Dial(SIP/${EXTEN}@sipgate-out)
Good luck,
Markus
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[asterisk-users] Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.
Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Geschenke Moritz: dunkle Schokolade. Geschenke Anna: normale Schokolade. Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Wed, 10 Dec 2014 05:12:16 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org + Haehnchenfleisch fuer Markus, kleine Kartoffeln/Pellkartoffeln/Ofenkartoffeln, + http://emmi-gerber.ch/index.php?id=1430 Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Wed, 10 Dec 2014 04:13:52 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Tanne + Schmuck Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: 26. Datum: Tue, 09 Dec 2014 14:21:23 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org ja ich hab nen Caquelon und einen Rechaud. Dazu n Brenner und noch genug Brennpaste. Das wars. Man braucht noch Teller, Gabeln. Und Käse, gerieben. Vom Stück, nicht aus der Tüte. Brot. WeiÃ~_brot oder Fonduebrot. WeiÃ~_wein. Kirschschnapps. Etwas Speisestärke. Knoblauch. Viel Pfeffer aus einer Mühle, frisch und schwarz. Beilagen: Mixed Pickles, Trauben, ggf Bündnerfleisch. Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: 26. Datum: Sun, 07 Dec 2014 01:57:50 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Panna Cotta Und dann kann man sie am 24.12. oder am 26.12. den Gästen servieren. Mit Erdbeeren aus der Kühltruhe zum Beispiel oder mit echter Ananas, Mango usw. Weitergeleitete Nachricht Betreff: Fwd: Fwd: 26. Datum: Fri, 05 Dec 2014 21:16:09 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Moritz - wie viele Spiesse / Zubehoer fuer wie viel Personen? Weitergeleitete Nachricht Betreff: Fwd: 26. Datum: Fri, 05 Dec 2014 21:12:04 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Anna: Fructoseallergie Weitergeleitete Nachricht Betreff: 26. Datum: Wed, 03 Dec 2014 23:20:09 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Moritz: http://www.chefkoch.de/rezepte/622761162560565/Schweizer-Kaesefondue.html + Salat -- _ -- 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] Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.
OMG.. how embarassing.. that was my personal reminder E-Mail for x-mas dinner. Not meant for this list. Please ignore. Shame on me.. *blushing* LOL. Am 12.12.2014 um 21:19 schrieb Markus: Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Geschenke Moritz: dunkle Schokolade. Geschenke Anna: normale Schokolade. Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Wed, 10 Dec 2014 05:12:16 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org + Haehnchenfleisch fuer Markus, kleine Kartoffeln/Pellkartoffeln/Ofenkartoffeln, + http://emmi-gerber.ch/index.php?id=1430 Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Wed, 10 Dec 2014 04:13:52 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Tanne + Schmuck Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: 26. Datum: Tue, 09 Dec 2014 14:21:23 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org ja ich hab nen Caquelon und einen Rechaud. Dazu n Brenner und noch genug Brennpaste. Das wars. Man braucht noch Teller, Gabeln. Und Käse, gerieben. Vom Stück, nicht aus der Tüte. Brot. WeiÃ~_brot oder Fonduebrot. WeiÃ~_wein. Kirschschnapps. Etwas Speisestärke. Knoblauch. Viel Pfeffer aus einer Mühle, frisch und schwarz. Beilagen: Mixed Pickles, Trauben, ggf Bündnerfleisch. Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: 26. Datum: Sun, 07 Dec 2014 01:57:50 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Panna Cotta Und dann kann man sie am 24.12. oder am 26.12. den Gästen servieren. Mit Erdbeeren aus der Kühltruhe zum Beispiel oder mit echter Ananas, Mango usw. Weitergeleitete Nachricht Betreff: Fwd: Fwd: 26. Datum: Fri, 05 Dec 2014 21:16:09 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Moritz - wie viele Spiesse / Zubehoer fuer wie viel Personen? Weitergeleitete Nachricht Betreff: Fwd: 26. Datum: Fri, 05 Dec 2014 21:12:04 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Anna: Fructoseallergie Weitergeleitete Nachricht Betreff: 26. Datum: Wed, 03 Dec 2014 23:20:09 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Moritz: http://www.chefkoch.de/rezepte/622761162560565/Schweizer-Kaesefondue.html + Salat -- _ -- 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] How to find RTP address of ongoing call?
Hi list, probably this is a FAQ but I can't seem to find it. How to find the RTP IP address of an ongoing SIP call? sip show channels seems to list the RTP address under the very left column called Peer. And it also lists the associated Call ID which I could associate with a call by executing sip show channel Call ID and before figuring out the Channel by running core show channels concise, but the issue is that the Call ID output from sip show channels is cut off and limited to 16 characters. Thanks! Markus -- _ -- 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] PBX hacked: why hundred of calls to the same number ?
Am 01.10.2014 11:40, schrieb Olivier: Some special numbers generate here and there revenues for callees (and not for callers). Not just some, but ALL numbers generate revenue for the receiving telecom. (Ok ok, a few exceptions, in the US for example) This is how telecoms have been earning money, ever have been and will for a while longer until interconnection fees for incoming traffic will be dropped completely, it's a work in progress, especially in the EU. (Unfortunately) There are 2 schemes: 1) Not so popular, but it's on the rise in the last 1-2 years: get landline numbers in country xyz, strike a deal with the telco that owns these numbers so that they'll pass a bit of their revenue on to you, and find a way to call yourself for free or at a lower rate than these numbers pay (= abuse your unlimited subscriber plan). The revenue is usually in the area of 0.00x or even 0.000x per minute, depending on the country. 2) Just google International Premium Numbers, or short, IPRN. It's a whole world of its own. Revenue is much higher. These are not real numbers and they never have full worldwide connectivity. So the fraudster has 2 tasks: 1) find a carrier through which he can reach these numbers and 2) find a way to call these numbers at a lower rate than they pay out. 2) is usually accomplished by hacking PBXes (= free calls), fraudulent apps etc. There are tons of stories of abuse regarding IPRN out on the web, just research a bit (quite interesting actually). Some technical background information on 1) How does it work? Where does the revenue come from you might wonder? First to be said, it can never work without a fraudulent telecom operator that is part of the scheme. Imagine you are calling from France to Latvia. Let's say the call passes France, Switzerland, Czech Republic and then goes to Latvia. Each carrier on the path passes the call on to the next carrier. Now, let's say the carrier in the Czech Republic is the evil one. The call comes in, and they simply say: well, this Latvian number that you just called belongs to us, we terminate the call here and pick it up. Billing time starts. Now, they charge the Swiss telco for the incoming call to Latvia, of course. And the Swiss telco charges the French telecom. The French telecom charges their subscriber (e.g. hacked PBX). The call never makes it to Latvia! Now, the Czech Republic telco works together with an IPRN provider (or they run an evil IPRN service by themselves kind of anonymously). They pass a bit of the money they get from the Swiss telecom on to the IPRN owner (the fraudster) and keep the remaining money for themselves. Easy money! This is why IPRNs don't have worldwide connectivity and can usually never get called from within a country (path is too short, no fraudulent telecom in between). They can even be real numbers that belong to someone, in this case, in Latvia, it doesn't matter. All you need to be is an evil telco where calls transit through and you have it. How much do you pay to your normal landline telco for a call to Latvia? To a Latvian mobile number? Let it be 0.25 EUR per minute. Thats what the subscriber pays, the Swiss telecom gets 0.22 of that, the Czech telco 0.20 and the fraudster 0.11. Just an example - margins are always high with IPRNs. Now you can simply do the same not with Latvia but with faaar away countries, islands (!) where calling to is even more expensive and your margins will go waaay up. Just to be clear: it's totally legit to earn money on incoming calls, this is the main income source for telcos all over the world. But abusing your unlimited plan and running IPRNs is not that legit I'd say. :) Beside sharing interests with the callee that get those revenues, why a hacker would like to dial the same numbers over and over ? I don't see another reason. In other words, in this case, is looking at callee number a promising path to find hackers ? Not in my experience. Since the fraudulent telcos work together with the IPRN owners you won't succeed. Must be a large-scale fraud scheme with millions of EURs lost for some authority to investigate properly. Plus, the IPRN owners even can get paid via Western Union etc. from the IPRN service, so all they need is a stolen/fake passport... so you are not left with much except maybe their IP address which, of course, if they are not totally dumb, isn't theirs. Gotta get in touch with some law enforcement agency and then catch them when they pick up the money at the Western Union counter. I should write a book about that. :P Cheers Markus -- _ -- 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
Re: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
Am 27.09.2014 17:28, schrieb d tbsky: can someone give an example for the function? thanks for the help. Not a programmer here, just grep -r'ed through the code, but maybe try one of these: G711A G711_ALAW -- _ -- 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] OT: Question on Caller ID (Spoofing calls with Asterisk)
Am 26.08.2014 16:45, schrieb Jeffrey Walton: I got a call from an overseas call center telling me about the problems with the Windows machine I was using. They wanted to remote in and fix things for me ... (Ignore the fact I use a MacBook Pro or an ASUS laptop with Debian). This is a common scam scheme. Try windows scam call at YouTube. Whether CLI spoofing works depends on the provider and has not so much to do with Asterisk. Asterisk makes it possible, but so does every common SIP softphone or most other SIP device for that matter. If the CLI you claim to own will get actually transmitted to the callee depends on the provider you are using for termination and whether they allow you to set an arbitrary CLI. It also depends on the routes that the call travels through until it reaches the callee. If there's a grey route on the path that overwrites your CLI, such as a GSM gateway, your CLI won't make it to the callee. Nowadays you can find providers that allow you set arbitrary CLIs on every corner. -- _ -- 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] Calls not hanging up
Am 08.08.2014 05:13, schrieb D'Arcy J.M. Cain: New data point - I just reverted to 11.10.2 without a single change to the configuration and the problem has gone away. Hmm. Could this have to do with session-timers (sip.conf)? I remember when I went from 1.4 to 10.7 I had to manually mess with the session-timers because my peers who delivered incoming calls would always end the call after 30 minutes. But your problem is kind of the opposite. :) Just a shot in the dark, without knowing much about SIP really, lol. If you really wanna know, you should fire up tcpdump and see what's going on there. -- _ -- 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] Attack on Sip server.
very simple, yet effective http://www.palner.com/blog/171/asterisk-no-matching-peer-found-block/ Am 27.06.2014 16:58, schrieb Steven Howes: On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com mailto:anuragrana31...@gmail.com wrote: There are lot of requests coming in and I am not able to stop it because I am unable to detect the IP address. I used wireshark to capture the packets. If you can capture the packet, surely you have the IP? If they intend to get the response then the IP header can't be forged. Steve -- _ -- 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] DTMF transmitting letter A
Dear list, maybe not really an Asterisk question, but... all my users dial in via PSTN (via SIP DIDs) and enter a target number via DTMF through my Asterisk 1.4. Out of about 150,000 calls per month I see on average about 1 call per month where an arbitrary caller enters the letter 'A' via DTMF. These callers use their mobile phones to dial in. I just reread the Wikipedia article on DTMF but I don't understand how someone can send an 'A'. Any clue? Thank you! Markus -- _ -- 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] High Availability with Asterisk
Hi Thorolf, Am 06.03.2014 16:21, schrieb Thorolf Godawa: Using (para-)virtualization with Xen could be an other option, on systems with low load this works reliable, but what happens on systems with high load? Are there any issues known about problems with the realtime, packet loss etc. because it runs in a VM? hmm, all my Asterisk'es run in (KVM) VMs, no issues there. But how is this related to high availability? I think it's not. :) I think the way to go for high availability (and scalability) is Kamailio! In a redundant setup, running on 2 separate physical machines (maybe in a VM, doesn't matter). Then you make them failsafe using whatever tool(s) available. Then you can set up 1, 2, 10 or 100 Asterisk behind Kamailio and any of them could fail (but 1 :-) ) and you will still be online. If you want to further develop the high availability thought, then you could use CephFS which will give you self-healing, 100% available storage over multiple physical storage servers. There you could store your Asterisk config files, or your MySQL database used by all the Asterisk servers, for CDRs, SIP registrations etc. It's kinda slow, but I think fast enough for Asterisk / MySQL. :) And, to scale and to make the Asterisk nodes redundant (redundancy is not really needed anymore, since Kamailio takes care of that, but basically then you get also VM/physical redundancy), you could look into OpenNebula which provides a nice auto-scaling feature already out of the box. If there's load on your Asterisk VMs, OpenNebula will detect this and spawn new Asterisk VMs (probably on different physical servers, otherwise it doesn't make that much sense performance-wise) which will automagically receive requests/calls from Kamailio. If the load goes down, the VM can be automagically stopped again to free resources for other VMs/applications. OpenNebula is less popular than OpenStack, which seems to be the first choice for Cloud-stuff today, but what I liked about OpenNebula is that it provides the auto-scaling feature already in the customer-facing web-frontend out-of-the-box, unlike OpenStack. So you could offer your customers a self-managed, redundant Asterisk cloud or something like that. :) In theory, this combination should give you a 100% redundant, auto-healing, auto-scaling VoIP setup. :) Regards Markus -- _ -- 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 as a client: can I get the remote SIP server to ignore rport?
Am 21.02.2014 15:12, schrieb Andres: Wow, if this is the case then I would be changing VM providers immediately.You would have problems not only with Asterisk but with most other services you wanted to host on it. There are many VM providers out there that work just fine with Asterisk even on a 1:1 NAT like Amazon Web Services. Update: the VPS provider has fixed the issue with their router and now everything works as expected. Unfortunately, in this particular exotic country, it seems there is exactly *1* VPS hosting provider, so switching to another provider would not be really feasible, and switching to another country is not at option either. :) All is good now! Thanks! Markus -- _ -- 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 configure asterisk to only accept SIP from kamailio@localhost but exchange RTP on all interfaces?
Am 20.02.2014 19:48, schrieb Alex Villacís Lasso: My concern is that asterisk is left listening for SIP through all interfaces and with no SIP passwords. I want to secure the setup against directed traffic to the asterisk UDP port (5080), that bypasses the kamailio process. I tried setting bindaddr=127.0.0.1 so asterisk will only listen for SIP traffic on localhost, but this has the side effect of also removing audio - the call appears to be successful on the softphone and on the asterisk logs, but no audio is actually heard. My theory is that the RTP traffic is being sent to kamailio instead of the softphone. Theories are nice, but you should check whether they are true using, e.g., tcpdump :) I would check with, for example: tcpdump -nnnqt -s 0 -A -i eth0 port 5060 or instead of port 5060 (or 5080) try udp to see what is going on with RTP. Change from eth0 to lo to see if there is really RTP going to nowhere. When looking at port 5060/5080, check the SDP header to see what kamilio/Asterisk/your softphone announce in terms of RTP. I thought kamailio is a SIP server/proxy only and is not involved in RTP at all. In any case, if you want to only allow only certain connections from somewhere to somewhere (including from/to certain ports), iptables is your friend if you are using Linux. -- _ -- 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 as a client: can I get the remote SIP server to ignore rport?
Am 20.02.2014 22:20, schrieb Rusty Newton: To force RFC3581 support for outbound REGISTER messages, you can set nat=force_rport in the general section of your sip.conf. (This also forces RFC3581 compliance for inbound messages, for any peers that inherit this general option) [...] Thank you for the suggestion, I just tried that. Unfortunately it doesn't help in my case. The problem remains that my hosting provider is rewriting ports: Packets on the hosted VM (1.1.1.1) where Asterisk client is running: IP 1.1.1.1.5060 2.2.2.2.5060: UDP, length 428 IP 2.2.2.2.5060 1.1.1.1.16025: UDP, length 544 IP 1.1.1.1 2.2.2.2: ICMP 1.1.1.1 udp port 16025 unreachable, length 556 And on 2.2.2.2, the remote SIP server (another Asterisk box for testing): IP 1.1.1.1.16025 2.2.2.2.5060: UDP, length 428 IP 2.2.2.2.5060 1.1.1.1.16025: UDP, length 544 IP 1.1.1.1 2.2.2.2: ICMP 1.1.1.1 udp port 16025 unreachable, length 556 The initial REGISTER received from 1.1.1.1 without force_rport: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK1780c521 And with force_rport enabled: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK279a0a9b;rport But of course, Asterisk on 2.2.2.2 will reply on the port where the request originated, in the above case, port 16025 which was rewritten from 5060 to 16052 by the evil hosting provider. This is what the Via header looks like when 2.2.2.2 answers to 1.1.1.1: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK279a0a9b;received=1.1.1.1;rport=16025 So, if there would be some kind of implementation that said Ignore the actual port that the request originated from, instead force answer on a port that is transmitted via some header setting, that would help me. Some fantasy option in sip.conf: nat_rport_forced=5060 Could generate a header in the initial REGISTER from 1.1.1.1 to 2.2.2.2 like this: Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK279a0a9b;rport_forced=5060 And now, even though 2.2.2.2 received the request originating from port 16025, it would honor the rport_forced setting and reply back on port 5060 instead of 16025. Maybe such an option exists, I just haven't found it yet? :) Thank you! Markus -- _ -- 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 as a client: can I get the remote SIP server to ignore rport?
Am 21.02.2014 01:33, schrieb Eric Wieling: To be fair NAT is rewriting your SIP packet source port. This happens all day, on almost every NAT device out there.Stop thinking it is purely a port rewriting issue, something else is going on. In the meantime, the provider has reconfigured the VM to work with the public IP address. That means the RFC IP address was removed and the public IP is now configured on the VM directly. The effect is the same, ports on outgoing packets still get rewritten. Have you set localnet and externip in sip.conf. Maybe the NAT device has a short UDP translation timeout -- try setting qualifyfreq=15 in sip.conf to generate traffic so the NAT box does not close the translations Yes, I have played around with local and externaddr. But the IP is not the problem, the port is. I think the translation timeout doesn't matter, because the router rewrites outgoing packets to a different port, but doesn't do so on incoming packets, and that's the issue. Here is an example sip show peers on one my my boxes. Three different locations are show. The ones you see with 5060 are either not NAT'd or they have a proxy at the customer location. The ones with a different port are NAT'd. And I'm pretty sure if you look at any of those peers that have a non-5060 port, the routers in front of them will rewrite packets destined for ports 53277, 4121, 47822 etc. to the proper corresponding internal IP:port where something is listening. The router of my provider won't. It rewrites ports on outgoing packets, but it passes incoming packets 1:1 to the VM. IMHO, my hosting provider is at fault, and I'm working with them to get it fixed. I was just wondering if there is some magic switch which can fix such a broken scenario. Thank you! Markus -- _ -- 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 as a client: can I get the remote SIP server to ignore rport?
Hi list, I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a client. I'm trying to SIP REGISTER with a remote SIP provider. The situation is that Asterisk is running in a VMware VM with a RFC IP address (192.168.1.2). The provider of the VM performs static NAT from the RFC IP address to a dedicated public IP address, however, they are rewriting ports at will. That's the problem. Here's an excerpt from tcpdump: IP 192.168.1.2.5060 my.provider.com.5060: UDP, length 411 REGISTER sip:my.provider.com SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK25a36d76 Max-Forwards: 70 From: sip:1...@my.provider.com;tag=as762d7322 To: sip:1...@my.provider.com Call-ID: 778c50f84e80a9db60dcd35a2f8a1498@127.0.0.1 CSeq: 228 REGISTER Supported: replaces, timer User-Agent: Asterisk PBX 12.0.0 Expires: 120 Contact: sip:1234@192.168.1.2:5060 Content-Length: 0 Then the remote SIP provider answers of course with 401 Unauthorized but that reply never makes it to Asterisk, because it doesn't come in on port 5060, where it actually originated on the VM, but on a random port that the VM hosting providers' NAT router rewrote to, in the below case port 63664. And the remote SIP provider tries to send the reply back on that random port. Note MY.PUBLIC.IP.ADDRESS and rport below: IP my.provider.com.5060 192.168.1.2.63664: UDP, length 534 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK25a36d76;received=MY.PUBLIC.IP.ADDRESS;rport=63664 From: sip:1...@my.provider.com;tag=as762d7322 To: sip:1...@my.provider.com;tag=as45cffa11 Call-ID: 778c50f84e80a9db60dcd35a2f8a1498@127.0.0.1 CSeq: 228 REGISTER Server: FPBX-2.10.0(1.8.15.1) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=1d46fec6 Content-Length: 0 I'm thinking the answer is no, but is there any option how I can get the remote SIP provider to answer me on port 5060? Without having them to change anything in their config. Thank you! Markus -- _ -- 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] Retaining P-Asserted Info
Am 16.02.2014 03:30, schrieb Nick Cameo:
Tried setting `sendrpid = yes` and still same problem. We really don't want to
have to `SipAddHeader` as it is already being formed by our switch.
From http://www.voip-info.org/wiki/view/Asterisk+SIP+trustrpid :
-snip-
P-Asserted-Identity
Asterisk does nothing when it receives a P-Asserted-Identity header. It
ignores it totally no matter what settings you use for trustrpid or
sendrpid. It does not copy it from an inbound call leg to an outbound
call leg for a bridged SIP-to-SIP call.
-snip-
I use SIPAddHeader to achieve what you described.
For example, I have a DID provider who only sends P-Asserted-Identity,
and at the same time a customer who needs that data as Remote-Party-ID,
so I'm doing this to send it to him:
exten = _X.,n,Set(RPID=${SIP_HEADER(P-Asserted-Identity)})
exten = _X.,n,Set(RPID=${CUT(RPID,\+,2-)})
exten = _X.,n,Set(RPID=${CUT(RPID,@,-1)})
exten = _X.,n,SIPAddHeader(Remote-Party-ID: ${RPID}
sip:${RPID}@my-ip-address\;privacy=off\;screen=no)
--
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[asterisk-users] Telco with multipe SIP servers
Hi! My telco is Deutsche Telekom and they got about 30 SIP servers right now. Currently I've set up a template for incoming calls in sip.conf and added each SIP server by it's IP address like this: [DTAG-in-1](DTAG-in-template) host=217.0.16.103 ... [DTAG-in-30](DTAG-in-template) host=217.0.20.99 I've done that to improve security and to be able to assign all calls coming in via Deutsche Telekom to a dedicated dialplan context. Unfortunately this approach is not scalable and it's a PITA to maintain a list of server IP addresses since Deutsche Telekom will get more SIP servers in the future. They've started to migrate the classic POTS/ISDN network to VoIP, the goal is get it done by 2016. Customers with DSL get VoIP directly, i.e. they need SIP phones or a SIP PBX, and customers with a phone line only are converted by the MSAN. And they don't provide an official list of the SIP servers :-( By some reverse engineering I found out that all SIP servers are within a specific subnet. Is there any way to match peers by subnet(s) instead of FQDNs or single IP addresses? If not, it would be a feature really needed to be able to cope with telcos running multiple or tons of SIP servers. cu, Markus -- / Markus Reschke \ / madi...@theca-tabellaria.de \ / FidoNet 2:240/1661 \ \/ \ / \/ -- _ -- 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 duration limit ? Calls end after 15 minutes...
Am 08.01.2014 16:07, schrieb Jonas Kellens: Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Look at session-timers in sip.conf. I had to set it to refuse for a specific provider because they are a little incompetent. Drawback is that a call can show as going on forever if the BYE message is lost due to network problems. -- _ -- 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 Mass exodus
Mike, Am 14.11.2013 01:48, schrieb Chris Bagnall: As in your case, it wasn't a connectivity or bandwidth issue - in the end we put it down to a disk I/O bottleneck. It might be worth running something like iostat on your box to see if you see a spike in iowait as voicemail is being checked. We resolved it simply by rate limiting our rsync jobs. In your case with a busy database, you might want to look at your MySQL indexes and/or cache settings - this might be something worth asking about on the respective MySQL discussion groups as well as here. you could also throw in some more stuff like the following and run it for 24 hours every second, then check the log for the timestamp to figure out what happened: while true; do echo `date`; asterisk -rx 'sip show peers'; asterisk -rx 'sip show channelstats'; mysqladmin --password=yourpass processlist; ps auxww; vmstat; iostat; echo; echo; echo; sleep 1; done log.txt Regards Markus -- _ -- 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] MusicOnHold starts magically for no reason
Dear list, on Asterisk 1.4.21 which is being used in a callthrough scenario - callers call via PSTN to a DID coming in via SIP and then dialing outbound via DTMF and the outbound calls get routed via some SIP termination provider - lately I see that every now and then MusicOnHold gets triggered like this on outbound calls: Started music on hold, class 'default', on SIP/outbound-sip-provider-0002 I do not have any reference to MusicOnHold in my extensions.conf so a misconfiguration is unlikely. Is there some SIP magic that can trigger MusicOnHold on my end? Thanks! Markus -- _ -- 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 high load when streaming MOH
Hi list, I've always about 50 concurrent SIP callers listening to several MOH streams (fed via mpg123) on Asterisk 11.4.0, 4x 2.2 GHz and the CPU usage is always at about 250% causing stuttering of the streams and delays when using Read() and Playback() (overall everything is just really slow/delayed). I've seen Asterisk doing that when the MOH stream that was fed via mpg123 was down... then it will just go nuts. But I'm pretty sure that all streams are active (checked via strace). My question is what would be the best way to look at the internals of Asterisk to really find out what exactly is causing that high CPU load? Thank you! Markus -- _ -- 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 high load when streaming MOH
Am 20.09.2013 15:17, schrieb jg: Sometimes I see something similar on my systems when several (3 or more) MOH streams are fed by mpg123. Suddenly some or all streams are stuttering, but the CPU load doesn't seem to go up significantly. My CPU load is permanent at 200-250%. I have 7 active mpg123 streams. On another box running Asterisk 10.8.0, 4x 2.4 GHz, I have 49 active streams, but they get fed via mplayer instead of mpg123. There the load is close to zero, but I have to say that I have only 1-2 concurrent callers there. Strange! Have you looked at what is happening with the receive queue of mpg123 (pgrep mpg123 and netstat -t -p)? netstat -t -p shows me each of the 7 IP addresses and Recv-Q has some bytes listed (55000 - 113000) for each IP. According to man netstat Recv-Q means The count of bytes not copied by the user program connected to this socket. Note the not. Now, is that good or bad? I have written a small daemon for myself that watches the receiving tcp streams and if nothing seems to be happening, simply touches musiconhold.conf and issues an asterisk -rx \moh reload\. Not nice, but it works, eventually. When I listen to the same music streams I find that they are not as stable as one would wish, at least at a time scale of several hours. What I do is simply call killall mpg123 every 10 minutes via cron. Asterisk MOH will restart the process immediately and there will be less than a second of silence for the listener. Not that elegant but it helps with hanging audio streams... When using several moh file classes, I have never hat audio problems. Me neither. Thanks! Markus -- _ -- 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 high load when streaming MOH
Am 20.09.2013 16:37, schrieb jg: My CPU load is permanent at 200-250%. I have 7 active mpg123 streams. I forgot something. Even if your 7 streams are mp3 streams they cannot consume the CPU power you are seeing. I thought so. So, I need to dig deeper... but how? Some command to show the different Asterisk internal sub-processes stats would be nice to begin with. Like what exactly is consuming that much CPU. -- _ -- 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] Macedonian DID
Am 04.09.2013 15:36, schrieb Zyumbilev, Peter: I searched a lot last few days but I am uanble to find a DID number in Macedoania. However no luck. any ideas about a provider ? didlogic.com had some a couple months ago, but they only lasted for a few weeks, probably offered by an individual and not a telco, then they were taken offline by the telco/regulator. I guess you'll have to wait a few years while this country progresses. But you can always get a premium number there (pay per minute/call) if that helps. BTW, asterisk-biz might be a better list for such requests. -- _ -- 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 use Skype ?
Am 03.09.2013 02:30, schrieb neo haux: I want to recieve calls to my Skype account and forward them to a SIP/FXS line. I searched for chan_skype for asterisk (v11), but found it only available for asterisk 10 I know that Digium gives no support for this module, but I am sure that someone somewhere did write some tool to allow such connectivity. Do have any idea if I can use Skype with my asterisk v11 ? I use SipToSis for this: http://www.mhspot.com/sts/ Asterisk version doesn't matter in this case. Anything that speaks SIP will probably work. -- _ -- 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] ReceiveFAX problem
Hi Matthias, Am 29.08.2013 23:17, schrieb Matthias Leopold: today i upgraded from asterisk 1.4.21 to 1.6.2.9 (i know this release is not supported anymore, please don't tell me to upgrade). unfortunately now i can't use the rxfax() application anymore. I don't know about rxfax/ReceiveFAX, but I'm using iaxmodem and HylaFAX to receive faxes and send them to different recipients depending on the extension that the fax was received on. I wrote a small how-to which you can find here: http://truemetal.org/universe/blog/2013/03/receiving-faxes-with-asterisk-iaxmodem-and-hylafax-with-dynamic-email-recipients/ HTH Markus -- _ -- 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] Is this application possible with Asterisk?
Hi list! Is the following application possible with Asterisk with the default functionality that is currently implemented? 1. User dials in via PSTN / SIP. 2. System announces random but individual PIN to user. 3. If there are no other users, user gets put through to a party by Asterisk dialing out via SIP and the user gets connected to that person. Let's call that person callee. 4. If the callee is already in a conversation, a new user (caller) gets put into a waiting mode (QUEUE?). Only 1 user can speak with the callee simultaneously. There is only a single callee, so there can always be only one conversation. 5. Each user has 15 minutes talk time with the callee. 6. Once the talk time is up, that user gets disconnected and the next user in the queue gets connected to the callee. (The user that is waiting the longest) 7. If a user in waiting mode should get disconnected for whatever reason, he can redial and enter the PIN to take back his previous position in the queue. 8. While in waiting mode, every user eavesdrops to the conversation between the callee and the currently talking user (CHANSPY and/or CONFBRIDGE LISTEN-ONLY). 9. If the user gets tired of eavesdropping on the conversation and to make his waiting time more entertaining, he can press a digit while in waiting mode to enter a conference that is going on between all users in waiting mode who entered the conference (CONFBRIDGE). 10. Difficulty: even when currently in a conference, a user must get automatically connected (or, at least, notified) to the callee when the callee is available now and the user is the next one in the queue. My doubts: Is queueing the right approach? I know there is a queue functionality in Asterisk but I'm not sure if it can be used for the above mentioned purpose. Can a user join a conference while being in the queue / in waiting mode? And when he exits the conference, can he take his old place in the queue again? How to send the user to the callee once the callee is available again and if the user should still be in the conference? Thank you so much! :) Regards Markus -- _ -- 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] Exceptionally long queue length (help!)
Help! I have providers configured that send me incoming calls via SIP. There's one that seems to make trouble. As soon as I get a few concurrent calls through this peer, Asterisk CPU load goes up to 250%, audio becomes laggy and I get hundreds of these per second in the logs: [Jun 20 20:00:16] WARNING[1206][C-0001] channel.c: Exceptionally long queue length queuing to SIP/provider11-0001 And the following is what I always get every few seconds for every call via this peer. I think this only happens when I'm answering the call in Asterisk itself, i.e. right now I'm just streaming MusicOnHold to callers. [Jun 20 20:00:22] NOTICE[1206][C-0001] channel.c: Dropping incompatible voice frame on SIP/provider11-0001 of format ulaw since our native format has changed to (alaw) Peer config [provider11] host=121.16.15.230 type=peer insecure=port,invite context=provider-in disallow=all allow=alaw canreinvite=no dtmfmode=rfc2833 Before I had allow=ulaw allow=alaw and removed the ulaw codec hoping that the above NOTICEs might disappear, but without success. Asterisk 11.4.0-1 from the official repository under CentOS 6.4 64bit. Help! :) -- _ -- 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] Integration with skype
Am 23.05.2013 15:14, schrieb Marie Fischer: For voice calls, you could try Skype Connect, which is SIP - but needs a business account, so not free. http://www.skype.com/en/features/skype-connect/ For voice, you can use SipToSis. Works flawlessly with Asterisk and the best part, it's free. :) www.mhspot.com/sts/ (site is down right now) -- _ -- 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] Integration with skype
Am 23.05.2013 16:04, schrieb Richard Kenner: For voice, you can use SipToSis. Works flawlessly with Asterisk and the best part, it's free. :) www.mhspot.com/sts/ (site is down right now) And that's related to the problem with it: it hasn't been maintained for quite a while. True, but it's still working if you follow the instructions carefully! :) If you know of another FREE alternative let me know. I think there is also something available for FreeSWITCH. The only problem with SipToSis that I found is that if you run it in a VM (KVM in my case), and have more than 1 concurrent call going on, the audio of the second call will drift away and become asynchronous after a few seconds into the call. Couldn't find a fix so I set up several VMs for multiple concurrent calls. :D If you use a dedicated server you probably won't have that issue... -- _ -- 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] On SIP INVITE answering to IP:port found in Contact: header.
Hi list! I'm trying to get a DID routed to me and the provider seems to have an unusual setup. Or maybe not? From looking at their SIP header they are using BroadWorks. The problem: they're sending their SIP invite from port 36252. My Asterisk 10.7.1 is answering to that port 36252 but their BroadWorks thingie is not listening on that port, but instead on port 5060. So they want me to send my SIP responses to the IP:port they send via the Contact: header. Here's the output from tcpdump (1.1.1.1 = them; 2.2.2.2 = me): IP 1.1.1.1.36252 2.2.2.2.5060: UDP, length 845 [...] Contact:sip:1.1.1.1:5060 [...] My Asterisk' reply: IP 2.2.2.2.5060 1.1.1.1.36252: UDP, length 602 [...] I tried the following in the peer config in sip.conf: host=1.1.1.1 type=peer insecure=port,invite port=5060 But that didn't change anything. How can I tell Asterisk to honor and use the info found in Contact: when sending its replies? Thank you! Markus -- _ -- 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] On SIP INVITE answering to IP:port found in Contact: header.
Joshua, Matthew, Am 17.04.2013 02:01, schrieb Matthew J. Roth: Joshua Colp wrote: If you set nat=no for that specific peer it should work as you need. 'rport' is forced on these days which works for most situations, except with some platforms and Cisco phones. _ Joshua, That sounds much easier than what I came up with, so I'd recommend to Markus that he try your suggestion first. bingo, that fixed it! Now everything's working fine, and my config looks like this: host=1.1.1.1 type=peer insecure=port nat=no Thanks a lot! Although I have to say I don't understand what is going on exactly. :) As can be seen below, and as Joshua suggested, nat=no removed the rport in the Via: header. I'm just wondering how Asterisk now knows to which port to send the replies to? Is it simply using 5060 because that's the RFC default for SIP? And because there is no port specified in the Via: header? Would the nat=no switch also fix it in a scenario where the remote side was sending from port 36252 but wanted the replies on port 5061 instead of 5060? If you have a moment, please take a look at my response and let me know if my understanding of the Contact and Via headers was correct. If it was, is the 'nat=no' solution just a way to workaround the provider's RFC-noncompliant platform? I would be interested in that, too. Matthew, to provide you with feedback, here are the SIP headers before and after nat=no: Without nat=no: IP 1.1.1.1.36252 2.2.2.2.5060: UDP, length 845 INVITE sip:1234@2.2.2.2 SIP/2.0 Via:SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKBroadWorks.2plg2e-2.2.2.2V5060-0-164693126-1529565735-1366128986378- From:sip:@1.1.1.1;user=phone;tag=1529565735-1366128986378- To:My wifesip:1234@2.2.2.2:5060 Call-ID:BW161626378160413-1212022685@1.1.1.1 CSeq:164693126 INVITE Contact:sip:1.1.1.1:5060 Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE Accept:application/media_control+xml,application/sdp,multipart/mixed Max-Forwards:40 Content-Type:application/sdp Content-Length:206 [ ... SDP removed ... ] IP 2.2.2.2.5060 1.1.1.1.36252: UDP, length 602 SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKBroadWorks.2plg2e-2.2.2.2V5060-0-164693126-1529565735-1366128986378-;received=1.1.1.1;rport=36252 From: sip:@1.1.1.1;user=phone;tag=1529565735-1366128986378- To: My wifesip:1234@2.2.2.2:5060 Call-ID: BW161626378160413-1212022685@1.1.1.1 CSeq: 164693126 INVITE Server: Asterisk PBX 10.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:1234@2.2.2.2:5060 Content-Length: 0 And with nat=no: The same, except for the reply: IP 2.2.2.2.5060 1.1.1.1.5060: UDP, length 588 SIP/2.0 100 Trying Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKBroadWorks.2plg2e-2.2.2.2V5060-0-179354203-809967599-1366158308532-;received=1.1.1.1 That means the rport part is gone. Regards Markus -- _ -- 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] VoIPGMap: Graphing active Asterisk calls on Google Maps
Hi list, I've always wanted to graph my active SIP calls on a map somehow, and now I've finally taken the time to do it. My script is called VoIPGMap and it displays active calls on Google Maps. Logic-wise it's designed for a callthrough or calling card scenario, where calls will be displayed to originate from a single location, but you can surely adjust it to instead show caller and callee location connected. I think the cool part is that it really tries to get the most accurate coordinates for every destination, down to the city level, by using text (country code, ISO, lat/long) and MySQL (ITU self-compiled DB + MaxMind public DB) based databases, which are included. Here are some demos: http://truemetal.org/universe/voipgmap/live_demo1.html http://truemetal.org/universe/voipgmap/live_demo2.html http://truemetal.org/universe/voipgmap/live_demo3.html http://truemetal.org/universe/voipgmap/live_demo4.html First I thought about calling it AstGMap but then realized you can simply feed it any form of E.164 numbers so I chose VoIPGMap instead :) If you feel like rewriting it to PHP or something like that for better performance, please do so. Unfortunately I'm not a programmer or I wouldn't have chosen shellscript. :) You can get it here: http://sourceforge.net/projects/voipgmap/files/ And if you like it and decide to use the output on the web somewhere, please let me know the URL so that I can check it out, thanks! Regards Markus -- _ -- 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] Moving User Agent To Remote Location
Am 03.01.2013 21:21, schrieb Nick Khamis: Oh that's so smart!!! So, if I did not misunderstand you, for this one call, have: rtpstart=10004 rtpend=1008 do you mean 1_000_8 ? Markus -- _ -- 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] Top Posting
Hi, one more hint... (trying to translate the commands to english) in Thunderbird open - Extras - Filter.. - Filter-Name: enter Top Posting Subject - Contains: enter Top Posting Action: Delete Markus Am 02.01.2013 21:31, schrieb Steve Totaro: On Wed, Jan 2, 2013 at 12:25 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 12:20 PM, Steve Totaro wrote: On Wed, Jan 2, 2013 at 12:00 PM, j...@millican.us j...@millican.us wrote: On 1/2/2013 11:30 AM, Richard Kenner wrote: If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. Ok folks, could not stop myself any longer. This pissing and moaning is foolish to say the least. There was a post a while ago in the original hijacked thread by Steve Edwards that gave a link to the rules of the list at: http://www.asterisk.org/community/discuss/ GO READ THEM! Directly before the list of Rules is: Show consideration. It's important to read the rules before posting on a mailing list. Sage advice if you ask me, and yes I know nobody actually asked me. It is not hard to follow the rules . If the nice folks at Digium took the time to post rules we should at least TRY to follow them. If you do not like the rules you can always petition Digium to change them but, taking up bandwidth on the list in this all to frequent pissing match is a futile waste of time. Grow up, follow the rules, have a good day. JohnM I became a list member way before any such rule and never had to click through and agree to these update ToS. I am grandfathered in. Thanks, Steve Totaro So Steve, can I steal this and send it to the IRS? The ATF? Local Police Department? G Wouldn't that be nice! Sorry couldn't resist. JohnM What the hell are you implying? The local police love me, I am in good standing with the ATF, FBI, DoD, DoS, USAID, DoE, DoL, and NSA. IRS wants some money in April but don't they always? LOL. Thanks, Steve T -- _ -- 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] Problem with SIP trunk I've set up between two * boxes.
Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call from box2 to box2?
Because local_SIP is the context on box2, and on box1 it's adhearsion.
The console message you pasted shows @local_SIP however, so it looks
like you are calling from box2 to box2?
Am 10.12.2012 22:53, schrieb Ken D'Ambrosio:
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
Re: [asterisk-users] SIP and RTP on different IP's
Am 25.11.2012 15:41, schrieb Tiago Geada: yes I have no control over that. Ok we will figure another way. Thanks How about two Asterisk instances with different bindaddr= settings? One instance listens on 172.16.1.10 and the other instance listens on 10.34.18.250. Maybe this will make the RTP IP in SDP automatically right? But I'm just guessing... Here are some instructions for multiple instances: http://forums.asterisk.org/viewtopic.php?f=1t=71510 Regards Markus -- _ -- 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] Intruder
Hi Felix, ngrep -W byline port 5060|grep -B1 INVITE sip Markus Am 16.11.2012 17:50, schrieb Ruben Rögels: Hi Felix, you have several things to check: netstat -a -n --udp --tcp will show you connections and connection attempts on network layer level. You have to look for incoming connections to port 5060 and if the call has been established for connections on your rtp ports. (see rtp.conf). If you can see connections not supposed to be there: thats your intruder ;-) I suggest you disable guest calls and you configure a default context in which dialed extensions can't be routed to charged destinations. sip.conf: allowguests=no defaultcontext=default extensions.conf: [default] exten = _X.,1,Answer() exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(ss-noservice) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,MusicOnHold(default,10) exten = _X.,n,PlayBack(silence/1) exten = _X.,n,PlayBack(vm-goodbye) exten = _X.,n,HangUp() The next step would be using fail2ban or something similiar to check the asterisk log for intruders. fail2ban recognized them and dynamically sets appropriate firewall rules. Good luck. best regards, Ruben Am 16.11.2012 17:20, schrieb Felix Vazquez: I am in the asterisk CLI and can see an unidentified caller trying the make calls out of the asterisk system. How do I stop them? How do I identify them and how can I see how the go in? This is an example of what I would see: NOTICE[4098]: chan_sip.c:20063 handle_request_invite: Call *from '' *to extension '90111235551212' rejected because extension not found. Felix This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient, be aware that any review, disclosure, copying, distribution or use of this transmission or its contents is prohibited. If you have received this transmission in error, please notify the sender immediately. -- _ -- Bandwidth and Colocation Provided byhttp://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] Restarting MOH
Hi list, I'm streaming live mp3 streams (web radios) via mplayer and mpg123 using MusicOnHold. Often, one or two of the streams die during the day and I have to restart (not reload) Asterisk to bring it back. moh reload on the console doesn't do anything. This is suboptimal if there are active callers, of course. Is there any way to restart the MOH system without restarting Asterisk? Asterisk 10.7.1 and 10.8.0. Thanks! Markus -- _ -- 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] Restarting MOH
Am 13.11.2012 16:51, schrieb OCEANET - Cédric BASSAGET: have you tried module reload res_musiconhold.so ? Hi Cedric, thanks for the suggestion. Unfortunately, it does nothing, just like moh reload. Any other suggestions? Regards Markus -- _ -- 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] Restarting MOH
Am 13.11.2012 19:01, schrieb Eric Wieling: module unload res_musiconhold.so and module load res_musiconhold.so Great, that works, but only if no caller is listening to MOH at that time. Since *all* my callers are listening to MOH and nothing else, that means for me it's the same like an Asterisk restart. When I try to unload the module I get: loader.c:542 ast_unload_resource: Soft unload failed, 'res_musiconhold.so' has use count 2 Is there a way to force the unloading? Any other suggestions? Thank you! Markus -- _ -- 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] Restarting MOH
hi, try to catch in in a cron job per minute. asterisk -rx 'module unload res_musiconhold.so' Markus Am 13.11.2012 19:15, schrieb Markus: Am 13.11.2012 19:01, schrieb Eric Wieling: module unload res_musiconhold.so and module load res_musiconhold.so Great, that works, but only if no caller is listening to MOH at that time. Since *all* my callers are listening to MOH and nothing else, that means for me it's the same like an Asterisk restart. When I try to unload the module I get: loader.c:542 ast_unload_resource: Soft unload failed, 'res_musiconhold.so' has use count 2 Is there a way to force the unloading? Any other suggestions? Thank you! Markus -- _ -- 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 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
Am 11.11.2012 11:46, schrieb Eric Kuhnke: I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: Miguel already explained what's going on. Have a look at the SIP packets to figure out more. On the Asterisk box: tcpdump -nnqt -s 0 -A -i eth0 port 5060 Also, check your router/firewall logs, respectively activate them, to find out why the packets are not going through. Maybe also try qualify=yes in your Diamondcard SIP stanza to help keep NAT open? Is this a D-Link router? -- _ -- 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] Web based Click to Call Application
Hi, I suppose WebRTC is the best solution nowadays, it's extremely interesting. I developed a C2C app in 2008, starting with call files and AMI, ended with asterisk-java and asterisk.NET to solve it. Hint: Try to solve (al)most (all) of your problems using Dialplans/Variables. Basically it's just one originate action using local channels. Markus Am 09.11.2012 11:38, schrieb Binan AL Halabi: Hi, Here is a starting point (WebRTC): https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support Regards. // Binan. *Från:* akhilesh chand omakhileshch...@gmail.com *Till:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Skickat:* fredag, 9 november 2012 11:32 *Ämne:* [asterisk-users] Web based Click to Call Application Dear All, I want to develop click to call(C2C) web based application.Is there any study material. I will really appreciate your help, thank you. Regards Akhilesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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
Re: [asterisk-users] Registering Asterisk to a SIP Provider
Am 16.10.2012 18:38, schrieb Sahil Gupta: The provider appears to be running PortaSIP. Anyone with suggestions? What does your register = line look like in sip.conf? (without the password) What does sip show registry show? What do you see on the Asterisk console (asterisk -vr) when you place a call to the account? -- _ -- 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] deny=0.0.0.0.0/0.0.0.0.0 does not seem to block external access
Am 01.10.2012 22:13, schrieb Eric Smith: I have the following in sip.conf for asterisk running on localhost. deny=0.0.0.0/0.0.0.0 permit=127.0.0.1/255.255.255.255 Really on localhost only, no peers other than from 127.0.0.1? Not sure why deny/permit is not working, but you could do bindaddr=127.0.0.1 in your sip.conf. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.10
Am 01.10.2012 20:43, schrieb motty.cruz: I can't find a clear procedure to lower musicOnHold volume! Any suggestions? exten = 1234,n,Set(VOLUME(TX)=-3) exten = 1234,n,MusicOnHold(default) in your extensions.conf should do the trick. (play around with the -3 value) -- _ -- 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
Am 29.09.2012 10:49, schrieb resea...@businesstz.com: [tz-ivr01 ~]# uptime 11:00:32 up 776 days, 10:49, 3 users, load average: 3.06, 3.05, 2.57 Sharing is caring Is that a Quad Core CPU in your box? PS: Yes, Asterisk is great. :) -- _ -- 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
Am 29.09.2012 20:17, schrieb Mitch Claborn: Sam - can you send output from a top when your server is under load? Just curious. Preferably with all CPUs showing (hit 1 in top) - 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
[asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI confbridge show profile user profilename. It's an all-SIP scenario with RFC2833 as the DTMF protocol. Is this a known bug? Thank you! Markus -- _ -- 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] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi Leif, Am 28.09.2012 13:24, schrieb Leif Madsen: Searching the issue tracker (hint, hint) does not return any dtmf_passthrough issues other than this one[0], which doesn't look to be related. thanks for your reply. Right, doesn't look related. Is another channel connected to the conference receiving the DTMF? Is that what you're intending? Because from my understand that is the intention, and not simply to limit the DTMF from being in the conference in the first place. At least that is almost how it reads in your message. All I'm trying to achieve is that the rest of the conference doesn't hear the beep's when a user presses a key. Users press keys to adjust the volume of the conference, for example. And these key presses get transmitted to all the other users in the conference, which can become quite annoying when there is a larger amount of users. Are you refering to my previous mails about adjusting volume of background music/speech in the conference? This is unrelated - in my test scenario I just set up a simple ConfBridge with no features at all, then dialed in via PSTN (arrives as SIP) from two different phones, and on each phone I can hear the key presses of the other party. OH! I just tested with a SIP softphone (X-Lite), and DTMF does not get passed to the other users! In X-Lite I can hear the DTMF keypresses of the users connected via PSTN (incoming via SIP), but when I hit a key in X-Lite I can't hear that on the PSTN phones. Hmmm ... Thank you! Markus -- _ -- 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] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Hi again Leif, Am 28.09.2012 13:42, schrieb Leif Madsen: OH! I just tested with a SIP softphone (X-Lite), and DTMF does not get passed to the other users! In X-Lite I can hear the DTMF keypresses of the users connected via PSTN (incoming via SIP), but when I hit a key in X-Lite I can't hear that on the PSTN phones. Hmmm ... I am not referring to your previous posts, but your test and results seem to indicate what I had somewhat thought. When you're using X-Lite, you're likely using RFC2833 for the DTMF method, which is out of band, and gets absorbed by Asterisk by it not playing the DTMF into the conference. This is how it should work (and likely does for most scenarios/phones). It sounds like maybe either a configuration or implementation issue on the carrier side though. Are you using inband DTMF there? Asterisk should really be absorbing that too, but sometimes it can't get it all. If you switch to an out of band method like info or rfc2833, does that help? Do you hear the DTMF on a normal call outside of ConfBridge() with the same carrier? I suspect this isn't a ConfBridge() problem, but a general DTMF one. Nice idea on the dtmf_passthrough setting, but it's not really the solution to your problem here. Thanks for the suggestions. (RFC2833 was already set on the SIP peer in question) For further debugging I've just done the following tests with these results: DID provider 1, incoming via SIP (Germany): PSTN to DID to X-Lite, RFC2833: hear DTMF, not logged on console PSTN to DID to X-Lite, inband: hear DTMF, not logged on console PSTN to DID to X-Lite, INFO: hear DTMF, not logged on console PSTN to DID to ConfBridge, RFC2833: hear DTMF, LOGGED on console PSTN to DID to ConfBridge, inband: hear DTMF, not logged on console PSTN to DID to ConfBridge, INFO: hear DTMF, not logged on console DID provider 2, incoming via SIP (UK and Netherlands): PSTN to DID to X-Lite, RFC2833: hear DTMF, LOGGED on console PSTN to DID to X-Lite, inband: hear DTMF, LOGGED on console PSTN to DID to X-Lite, INFO: hear DTMF, not logged on console PSTN to DID to ConfBridge, RFC2833: hear DTMF, LOGGED on console PSTN to DID to ConfBridge, inband: hear DTMF, LOGGED on console PSTN to DID to ConfBridge, INFO: hear DTMF, not logged on console DID provider 3, incoming via SIP (Nigeria and Kazakhstan): PSTN to DID to ConfBridge, inband: hear DTMF, LOGGED on console DID provider 4, incoming via SIP (US): PSTN to DID to ConfBridge, RFC2833: hear DTMF, LOGGED on console X-Lite 3.0 to ConfBridge: X-Lite peer is set to dtmfmode=rfc2833 in Asterisk. X-Lite force inband YES, RTP 2833 YES: don't hear DTMF, LOGGED on console X-Lite force inband NO, RTP 2833 YES: don't hear DTMF, LOGGED on console X-Lite force inband YES, RTP 2833 NO: hear DTMF, not logged on console X-Lite force inband NO, RTP 2833 NO: don't hear DTMF, LOGGED on console The last result is kinda strange. Hm. PSTN means that I've tested two times, from a regular landline and from a mobile. Always calling to the providers DID which ends up in Asterisk via SIP. In the case of ConfBridge there were always 2 participants in the conference so that I could check if I hear the DTMF on the other end. not logged on console means that I can hear the DTMF tones in X-Lite/ConfBridge but Asterisk doesn't seem to recognize them (which is fine as not all providers support all DTMF variants). My resume is: DTMF is just fine, ConfBridge dtmf_passthrough is not working at all. Agree? :) Thank you, Markus -- _ -- 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] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
in sip.conf: input 123, output hear DTMF (123) X-Lite, INFO in sip.conf: input 123, output hear DTMF (123) (X-Lite tests only with force inband YES, RTP 2833 YES) If I understand right, all my four DID providers are broken? Thank you! Markus -- _ -- 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] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?
Am 28.09.2012 17:33, schrieb Joshua Colp: Ok, good idea, here are the results of Read() and SayDigits(): snipped results to make this email manageable How are you changing the DTMF for each provider? If you are merely changing it using dtmfmode in sip.conf this may or may not change how the provider side sends it. In the case of setting it to rfc2833 it causes RFC2833 to be negotiated in the SDP. Some equipment MAY change to using inband if it has not been negotiated. Yes, only via dtmfmode in sip.conf. I have no control over the provider side, of course. :) Or are there other options? Essentially when doing conversion of inband DTMF to out of band DTMF it is possible for some parts of tones to get through unmuted. You have to strike a balance between detecting the DTMF early enough, not detecting other stuff as DTMF, and muting it. Some implementations may let some leak through. The only way to completely overcome that is to buffer enough audio and delay the stream. Ah! So because I'm calling from PSTN which is naturally inband (I guess?) my DID provider (or rather some provider in front of him) converts to out of band (RFC2833), but the conversion cannot be fully accurate? If I understand right, all my four DID providers are broken? If the provider is doing the conversion their equipment should mute the inband DTMF as best it can and you should not hear it. How much of the tone are you hearing? About the tone itself: it's always different for each DTMF method/provider I would say. Sometimes it's just a really short glub (no idea how to describe that better), and sometimes it's a full-length beeep. But in any case except for provider 4 inband + INFO I'm hearing all of it (123), so it's anything from glub-glub-glub to beeep-beeep-beeep. :-) Right now, after what you wrote, I'm left with the feeling that RFC2833 is pretty much, well, hmm... useless in a PSTN-VoIP scenario? It works fine when using X-Lite, I suppose because there is no conversion involved and therefore the tones can get completely muted. I'm sad now. Is there nothing I can do to remove DTMF tones from my conferences? :-( Thank you! Markus -- _ -- 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] FAX via Asterisk
Am 27.09.2012 08:15, schrieb Shanavaz E A: Patrick, can you please give the steps to configure fax with iaxmodem and hylafax. Is it free to use? I'm not Patrick, but here's a good how-to that worked perfectly for me. In German, but it's basically enough just to type the commands :) And yes, it's free to use. http://das-asterisk-buch.de/faxserver-mit-iaxmodem-und-hylafax.html Regards Markus -- _ -- 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] FAX via Asterisk
Am 26.09.2012 17:53, schrieb Mark Robinson: I have asterisk 1.8.18 which connects to ISDN PRI. All phones are sip, Aastra 6757i. Everything works as expected. We also have a FAX machine. We need to be able to use that FAX machine to send or receive faxes. We are planning to have a dedicated did for faxes. Before, FAX machine was connected directly to pots line. Any digestions how to accomplish it? a) Without Asterisk: Get a ISDN-ATA (POTS-ISDN converter), e.g. DeTeWe TA 33clip, hook the fax machine to it and the ATA (configured to receive calls for the fax DID) to the ISDN line. At least thats how it would work with ISDN BRI, not sure about PRI. If you don't have enough ISDN ports on the NT you could get an ISDN switch. b) With Asterisk: Get a SIP-POTS adapter, configure it with the fax DID and hook your fax machine to it. There are many adapters available: http://www.voip-info.org/wiki/view/Analog+Telephone+Adapters (FritzBox seems to be popular) -- _ -- 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 crashing when recording ConfBridge calls (10.7.1)
Hi list,
since I enabled call recording in my ConfBridge bridge profile, after
about every 5th held conference (sometimes after 3, sometimes after 10,
it depends), when the last users exits the conference, Asterisk crashes.
Is this a known error? I'm using Asterisk 10.7.1 on CentOS 64bit with
the official RPMs.
lb1*CLI
[Sep 10 15:06:18] WARNING[21582]: file.c:496 filehelper: File
/var/spool/asterisk/confbridge/confbridge-name-22*68*39*09-1347282201.46.sln
detected to have zero size.
-- Bridge/0xaf10ba8-input Playing
'/var/lib/asterisk/sounds/chatfire_konferenz_verlassen_6.slin' (language
'en')
-- Bridge/0xaf10ba8-input Playing 'confbridge-leave.gsm'
(language 'en')
== MixMonitor close filestream (mixed)
== End MixMonitor Recording
ConfBridgeRecorder/conf-22*68*39*09-uid-1446923861
lb1*CLI
[Sep 10 15:06:19] WARNING[21582]: channel.c:2880 ast_hangup: Hard hangup
called by thread 1115089216 on Bridge/0xaf10ba8-output, while fd is
blocked by thread 1115597120 in procedure ast_waitfor_nandfds!
Expect a failure
lb1*CLI
Disconnected from Asterisk server
Executing last minute cleanups
And here's the snippet from confbridge.conf:
[chatfire-internal-bridge]
type=bridge
record_conference=yes
sound_has_joined=/var/lib/asterisk/sounds/chatfire_betritt_konferenz_6
sound_has_left=/var/lib/asterisk/sounds/chatfire_konferenz_verlassen_6
sound_only_person=/var/lib/asterisk/sounds/chatfire_erster_anrufer_6
sound_there_are=/var/lib/asterisk/sounds/chatfire_es_befinden_sich_derzeit_6
sound_other_in_party=/var/lib/asterisk/sounds/chatfire_anrufer_in_der_besprechung_6
sound_get_pin=/var/lib/asterisk/sounds/chatfire_intro_6
And the dialplan:
; internal conference only
exten = 06,1,Answer()
exten = 06,n,Set(CHANNEL(language)=chatfire)
exten = 06,n,Set(CONFBRIDGE(user,template)=chatfire-internal-user)
exten = 06,n,Set(CONFBRIDGE(bridge,template)=chatfire-internal-bridge)
exten =
06,n,Set(CONFBRIDGE(bridge,record_file)=/tmp/chatfire-${STRFTIME(,,%C%y%m%d%H%M)}.wav)
exten = 06,n,ConfBridge(22*68*39*09,,,chatfire-internal-menu)
exten = 06,n,Hangup()
Help! :)
Thank you,
Markus
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[asterisk-users] MusicOnHold (stream) interrupted after DDoS / network issues
Dear list, sometimes my network receives a large DDoS (packet flood, attack) that I manually block after a few minutes or it just stops by itself. As soon as the attack stops, the network is fine again, ping is back to normal, traffic levels fine again, enough bandwidth etc. During the DDoS a MP3 stream that I listen to via MusicOnHold will become interrupted, like music-silence-music-silence and so on, which is normal because of the DDoS and packet loss. However, the stream does not return back to normal after the DDoS is finished. Even minutes or hours later, its either still interrupted like described above, or there is complete silence. I'm calling mpg123 like this: /usr/bin/mpg123 -q -r 8000 -b 0 --mono -s http://stream.url I tried to kill the mpg123 processes (there are different streams) and also the shell script that calls them from MoH, but that doesnt change anything, although the processes get restarted. The only thing that helps is a total Asterisk restart, so I suppose the issue is within Asterisk in some strange way. Isn't that strange? Does anyone have an idea? Thank you! -- _ -- 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] ConfBridge announce_join_leave custom recording?
Hi list, I couldn't find any ConfBridge option to specify a custom recording to be played for announce_join_leave (After the tone, say your name and then press the pound key)? Is that not possible or am I missing something? As a workaround, I could probably specify another language and replace the vm-rec-name file? Thanks! Markus -- _ -- 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
