Re: [asterisk-users] Compiling Dahdi (Can't read private key)
Since a couple versions back I keep getting these messages when compiling Dahdi: make[2]: Entering directory `/usr/src/kernels/3.10.0-327.13.1.el7.x86_64' INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_dynamic.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_dynamic_eth.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_dynamic_ethmf.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_dynamic_loc.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_echocan_jpah.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_echocan_kb1.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_echocan_mg2.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_echocan_sec.ko Can't read private key INSTALL /usr/src/dahdi-linux-complete-2.11.1+2.11.1/linux/drivers/dahdi/dahdi_echocan_sec2.ko Can't read private key ... Anyone know what they are about? You can ignore this. I guess your system is CentOS 7. I used to know what the reason was, but forgot it. I guess when you google the message, you'll find an answer. jg -- _ -- 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] Phone Number Validation
Hi Everyone, I need to develop a service which tells me whether a given phone number is in service and is valid or not. It can be international number. This is basically to clean the list of leads we have. Is there any service which can give me the required information? I currently have an international numbering plan database which only tells me if the given phone number is in valid format up to a certain area code. But I need to know whether it will ring or not. Any help will be appreciated. Hi! I am doing something similar. Country codes are available from ITU-T. Country codes are available for every country, except for the North American Numbering Plan, which covers essentially North America. NANP numbers have a simple structure (with little oddities), which is not generally valid outside their domain, so it is difficult to check the validity of numbers (unless you are willing to work through the regulations of every country you want to cover). For example, a complete German phone number, including the equivalent of NPA and NXX, can be between 5 and 15 digits. The system is (almost) strictly hierarchical, but requires detailed knowledge, i.e. you do need an algorithm that figures out the area code. There are also separate number ranges for mobile phone numbers. In practice there can be more than 15 numbers, depending on the country, and whether the regulators are not particularly strict in enforcing a specific length of phone numbers (for ISDN lines). Generally, you cannot know whether dialing a number will ring the other end, or not. If all channels are already occupied for a T1 or E1 connection, the last exchange station will already signal unavailability, i.e. "user busy" may be signaled by the user or the network. jg -- _ -- 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] Dial()-Function
Hi all! :) I search a function or option for application Dail(). My situations: I have two or more Dial()s with multiple devices (Handgroups). Level1: Dial(SIP/device1,20) Level2: Dial(SIP/device1/device2,20) Level3: Dial(SIP/device1/device2/device3,20) When in level one, no one accept the call until the timeout, they have a missed call on device. When in level two, no one accept the call until the timeout, they have a missed call on device again. If SIP/device3 accept the call, SIP/device1 has two missed calls and SIP/device2 has one missed call. If on the same level anyone accept the call, the other in the same level get "Call complered elsewhere". (That's okay) If i use option "c" for Dial() in any case asterisk send "Call completed elsewhere". Also if the Caller hangup during ringing/cancel the call. What i need: On timeout: "Call completed elsewhere" (this is with option "c") If any other in the same level accept the call: "Call completed elsewhere" (Thats normal) And special, if the caller cancel the call during ringing: "Missed Call" (This is without option "c") But i need this behavior with option c, cause on timeout i need a "Call completed elsewhere". How can I achieve this? Sincerely, Dominique Wouldn't it be easier to use a local channel and do something like is done in the "Delay Dialing Devices Example"? https://wiki.asterisk.org/wiki/display/AST/Delay+Dialing+Devices+Example jg -- _ -- 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] Dial()-Function
No, i think unfortunately it is not easier. :/ I have a string from database (Macro/appdata) in the format: function|timeout|function|timeout|function|timeout| Up to seven value pairs. "function" can be "Queue" (Identified by: "qu"-string), "Voicemail" (Identified by: "vm"-string), "Anouncement" (Identified by: "an"-string), "Enddiveces" (Identified by: "SIP/"-string)) or an "external Number". Every function with an timeout to the next. I loop all. I have no idea how I can pass the function and the timeout to the extension by the most beautiful way. Without a variables war. One possibility would be to package the parameters in the extension, but that would be very ugly. Yes, today we would solve the most different. :) I can't see what you are trying to do and how your "appdata" relate to your previous mails. I am also wondering why you want to "pass" functions and timeouts. Wouldn't it be enough to dispatch everything, set some channelvars, assemble a dial string, and then let the local channels take care of the rest? jg -- _ -- 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] Voicemail issue on Grandstream GXP2000 phones
: From Line 3, it does not recognize the password. Did you check whether you have the same DTMF settings for Line 3? -- _ -- 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] Dial command: channel type detection
Some of my users connect to my asterisk box using SIP, other using iax
(in users.conf, I set "hasiax=yes" for those users).
How do I detect which protocol some user is using ? I cannot find any
variable which contains that information.
Reason is: I need this information for the Dial() command to work with
all my users, as the protocol is needed when using this command.
Why can't you evaluate the CHANNEL variable with something like
Set(TECHNOLOGY=${CUT(CHANNEL,/,1)})? One could also initially use a special context for IAX
channels and set a variable. It depends on what you want to do.
jg
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Re: [asterisk-users] Forwarding call if extension busy
Hi and happy new year!
My question:
- two extensions: and
- an active call on
- incoming calls to should be forwarded to (call advice!) and
I know how can I forward an incoming call to more than an extension, but I have no idea how
can I get the information, that has already an active call...
I think, I need something like:
exten => _,1,Verbose(2,Incoming call for - [${CALLERID(num)}])
exten => _,n,GotoIf( ?busy)
exten => _,n,Dial(SIP/,19,RcxX)
exten => _,n,VoiceMail(,us)
exten => _,n,Hangup
exten => _(busy),n,Dial(SIP//,19,RcxX)
exten => _,n,VoiceMail(,us)
exten => _,n,Hangup
Well, the problem is the second line, of course...
Of course the extension is NOT "really busy", since the phone can support more active
channels, but I hope I explained my problem...
Any suggestion?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
There may not be a general solution as the end points can accept more than a single call
themselves as described by yourself, i.e. the phone may not be in a busy state unless the max.
number of calls has been reached or a call has been actively rejected. In that case you might
put another Dial just after the first Dial application. If there is still no answer, VoiceMail
gets called.
You need to configure your phone to accept only a single call. Another approach would be to
check from within Asterisk whether a particular endpoint has already active calls and Dial() as
required, i.e. one would delete the phones with active calls from a given list. Since there is
no real "busy" condition, this seems to be a cleaner approach.
At first you should be able to describe exactly which behavior you want.
jg
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Re: [asterisk-users] Which router/firewall would you use for a virtual-PBX Asterisk installation?
Hi everyone. We've got a fairly large base of customers who use our Asterisk server for phone service in a virtual PBX kind of way, where the server is security hardened and exposed to the internet for them to connect to remotely with SIP and IAX. It's certainly not the sort of affair where we're running it as a PBX just within the building. As a result, we see network traffic coming through eth0 between 512 Kbps and about 3.0 Mbps, depending on the time of day. We haven't so far been using a hardware firewall/router on our server network, but it's becoming increasingly clear that we need to. We have enough experience to know that Asterisk is pretty sensitive when it comes to network hardware in our situation - we've had to replace one otherwise perfectly good 100 Mbps network switch because it simply wasn't able to keep up with the amount of streaming audio we put through it, and it badly affected voice quality. We have other traffic flowing through our server network too, including a significant amount of e-mail and web traffic, although that's not quite as sensitive to the quality of our network hardware. If you've got these large requirements for Asterisk, I'd love to hear what you use for a router, and whether that router has met your needs. It would also be nice to hear about what kinds of routers to avoid that you may have tried in the past and found lacking. I am working at a scale of about 10 Mbps and I am using customized pfSense setups. Essentially, I am also using Asterisk as a session border controller as part of the router/firewall. I am using a multi step procedure to keep unwanted traffic away from the application software, which includes geo IP filtering and blocking based on Snort alarms. So far I haven't seen the necessity to block anything based on Asterisk logs, but I'll plan to add that feature to pfBlockeNG as a custom IPv4 (and IPv6) list. It's too early for recommendations or public demo software, but I am planning to add my SBC to pfSense 2.3 superseding the current Asterisk package. If necessary, pfSense allows for traffic shaping and a couple of other neat feature, that are usually not part of small firewalls. jg -- _ -- 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] No sound with internal calls depending on which phones
Am 12.11.2015 um 16:22 schrieb (lists) Denis BUCHER:
Dear all,
I have a very strange problem :
* external calls work perfectly,
* internal calls between some phones too,
* but internal call between two similar phones don't work !!! (Snom 710)
When we have sound, there are no errors in asterisk. When we do not have sound, there is the
following error :
* [Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module
loaded, can't
setup SRTP session.
This is a working internal call :
== Using SIP RTP CoS mark 5
-- Executing [301@local:1] Dial("SIP/dbucher-", "SIP/phone1") in
new stack
== Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 is ringing
-- SIP/phone1-0001 answered SIP/dbucher-
-- Remotely bridging SIP/dbucher- and SIP/phone1-0001
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.99:49646 (type 00, len 000160)
Got RTP packet from192.168.128.99:49646 (type 126, seq 031575, ts 01,
len 00)
[Nov 10 17:50:50] NOTICE[21513]: res_rtp_asterisk.c:2190 ast_rtp_read: Unknown RTP codec 126
received from '192.168.128.99:49646'
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:57818 (type 00, len 000160)
== Spawn extension (local, 301, 1) exited non-zero on 'SIP/dbucher-'
This is a non-working call :
== Using SIP RTP CoS mark 5
[Nov 10 17:51:47] ERROR[21480]: chan_sip.c:28306 setup_srtp: No SRTP module loaded, can't
setup SRTP session.
-- Executing [301@local:1] Dial("SIP/hsolutionspf5-0002", "SIP/phone1")
in new stack
== Using SIP RTP CoS mark 5
-- Called phone1
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 is ringing
-- SIP/phone1-0003 answered SIP/hsolutionspf5-0002
-- Remotely bridging SIP/hsolutionspf5-0002 and SIP/phone1-0003
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.228:65494 (type 00, len 000160)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
Sent RTP P2P packet to 192.168.128.231:51350 (type 03, len 33)
== Spawn extension (local, 301, 1) exited non-zero on
'SIP/hsolutionspf5-0002'
I tried many options to disable SRTP but without success :
* canreinvite = no
* canreinvite = nonat
* srtpcapable=no
* encryption=no
* directmedia=nonat
* ...or noload => res_srtp.so in modules.conf
Any help would be GREATLY appreciated !
Denis
P. S. We have Asterisk 1.8.4.4 under CentOS release 5.11 (Final)
Please check
http://wiki.snom.com/wiki/index.php/Settings/user_srtp
and make sure the flag is off.
If you install Asterisk with the srtp module, then you need to set the auth-tag to AES-80, but I
haven't played with this option for quite some time.
jg
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Re: [asterisk-users] Find me macro - calling multiple people to get a hold of one
We're trying to set up a phone number that customers can call to get a hold of
anyone of a group of sysadmins (and not their voice mails!). We found the
findme example ([1]) that makes the callees press 1 to accept the call. It
almost works, but it doesn't work correctly when one of the callees, the
sysadmins, hangs up after accepting the call.
We're using this 'screen' macro:
==
[default]
exten => _XX,1,Dial(SIP/bla/${EXTEN:4},40,M(screen))
exten => _XX,2,Hangup
[macro-screen]
exten => s,1,Wait(1)
exten => s,n,Background(press-1)
exten => s,n,WaitExten(10) ; the value is the Wait time before we assume the
call is not accepted
exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to
connect the caller
exten => t,1,Playback(weasels-eaten-phonesys) ; if you're too late with
pressing 1
exten => t,n,Set(MACRO_RESULT=CONTINUE)
[findme]
exten => s,1,Set(CALLERID(all)="Alarm" <911>)
same => n,Playback(please-wait-connect-oncall-eng)
same => n,Dial(LOCAL/${WIEBE_MOBILE})
same => n,Playback(vm-nobodyavail)
exten => t,1,Playback(vm-nobodyavail)
=
First of all, what is MACRO_RESULT? I can't seem to find anything about that.
Googling for it yields basically nothing.
But the biggest problem is when the callee answers, then hangs up. The person
calling is connected to the phone that hangs up, instead of hearing
'vm-nobodyavail'. This seems to be because there is nothing that sets
MACRO_RESULT in that event (it's only set on 't', timeout).
I tried adding:
exten => h,1,Verbose(0,"The callee hung up")
exten => h,n,Set(MACRO_RESULT=CONTINUE)
to handle the hangup (h), but it's not doing that.
WaitExten() pushes the result back on the stack and restarts the context,
right? So what is the result when the person hangs up?
Regards,
Wiebe
Sorry, but why is a simple
Dial(SIP/A/B&...,${CALLTIMEOUT},${DIALOPTS})
...
Hangup()
not acceptable? If necessary, one can try to find out which devices are technically available to
avoid dialing a non-existent device. If pressing a "1" is acceptable, then why not pressing the
"DND" to not accept the call?
jg
There's
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Re: [asterisk-users] Why I get repeat messages many times
I am using the asterisk 13 and I config my dialplan for the SIP messaging as
the following :
http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()
With this configuration I could send message, but I don't know what wrong with it as sometimes
I get the repeat messages many times. do you have any idea?
Are the calls answered before jumping to astsms?
jg
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Re: [asterisk-users] Live Recording on the NAS?
I am planning to move Asterisk from physical server to a VM on a ESXi host. VMware datastore / VM's will be stored on the shared storage on the NAS (NSF). I might get Synology NAS. Do you store call live recording on the NAS? There would be around 60 concurrent calls recording at the same time and it may cause network bottleneck. There will be other VM's stored on the NAS like Windows Servers, Linux Servers, Database, etc. 60 concurrent alls sounds like a lot. I'd work with a RAM-disk and some post-processing to be safe. I have a low priority background task that moves finished sound files to a file server and converts them to mp3. The software that accesses the audio looks for both formats at both places. I think it is generally a good idea to handle file issues outside of Asterisk. jg -- _ -- 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 warnings no samples for alawtolin
[Aug 11 21:57:14] WARNING[1992] translate.c: no samples for alawtolin [Aug 11 21:57:14] WARNING[2005] translate.c: no samples for alawtolin [Aug 11 21:57:15] WARNING[2027] translate.c: no samples for alawtolin Hi to all, I have an elastix box running asterisk 1.8.20 without problem. It's about four days I've started seen in log a warning message saying translate.c: no samples for alawtolin, and now the frequency of this message is about 6 times a second. There's no other clue, everything is running smoothly and googling for it doesn't help. Here's an excerpt: [Aug 11 21:57:15] WARNING[2029] translate.c: no samples for alawtolin [Aug 11 21:57:15] WARNING[2038] translate.c: no samples for alawtolin [Aug 11 21:57:15] WARNING[2045] translate.c: no samples for alawtolin [Aug 11 21:57:15] WARNING[2055] translate.c: no samples for alawtolin [Aug 11 21:57:15] WARNING[2059] translate.c: no samples for alawtolin [Aug 11 21:57:15] WARNING[2078] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2093] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2095] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2110] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2120] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2125] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2132] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2139] translate.c: no samples for alawtolin [Aug 11 21:57:19] WARNING[2141] translate.c: no samples for alawtolin [Aug 11 21:57:20] WARNING[2152] translate.c: no samples for alawtolin [Aug 11 21:57:20] WARNING[2174] translate.c: no samples for alawtolin [Aug 11 21:57:20] WARNING[2177] translate.c: no samples for alawtolin [Aug 11 21:57:20] WARNING[2208] translate.c: no samples for alawtolin [Aug 11 21:57:20] WARNING[2210] translate.c: no samples for alawtolin [Aug 11 21:57:20] WARNING[] translate.c: no samples for alawtolin Does anyone have an idea of what is means and how I can get rid of it? Thanks AFAIK this is related to the settings of silence suppression. I haven't seen this for a while, but you might want to check the Silence Suppression, or Voice Activity Detection (VAD) settings of your SIP endpoints. jg -- _ -- 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] compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes
I have 2 strange errors when using the Background()-application and DTMF-input
that is received.
First of all, my first 2 lines are not being executed. The first line being executed is the
Set() application, thus line 3.
Secondly, the received digits (911) is not the same as the EXTEN (which is set
to 91).
exten = ivr,n,Set(TIMEOUT(digit)=2)
exten = ivr,n,Background(/var/lib/asterisk/sounds/${ASTPROMPT})
exten = _X.,1,NoOp()
exten = _X.,n,NoOp(input=${EXTEN})
exten = _X.,n,Set(choice=${EXTEN})
[Aug 7 12:31:26] -- Executing [ivr@pbx-routing:7] Set(SIP/SipAgenT-0626,
TIMEOUT(digit)=2) in new stack
[Aug 7 12:31:26] -- Digit timeout set to 2.000
[Aug 7 12:31:26] -- Executing [ivr@pbx-routing:8] BackGround(SIP/SipAgenT-0626,
/var/lib/asterisk/sounds/5003) in new stack
[Aug 7 12:31:26] -- SIP/SipAgenT-0626 Playing
'/var/lib/asterisk/sounds/5003.slin'
[Aug 7 12:31:41] NOTICE[3886]: ast_expr2.y:763 compose_func_args: argbuf
allocated 4 bytes;
[Aug 7 12:31:41] NOTICE[3886]: ast_expr2.y:782 compose_func_args: argbuf uses
3 bytes;
[Aug 7 12:31:41] -- Executing [911@pbx-routing:1] Set(SIP/SipAgenT-0626,
choice=91) in new stack
I have reloaded the dialplan several times, but the first 2 lines never get executed. In stead
they generate the error : ast_expr2.y:763 compose_func_args: argbuf allocated 4 bytes;
Anyone know what is going on here ?
Can you post the complete output with set verbose = 3? I think you didn't show all the code
that got executed. Also, Asterisk might not get what was dialed. If you have a phone with dial
plan settings and there is a regex which submits immediately after 2 digits for certain
patterns, you'll never get the complete number.
Given the code, there is no reason to execute the ivr extensions.
jg
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Re: [asterisk-users] PTT push to talk solution
This is for a government type end user. They wish to be having an internal meeting and be able to announce something - but require a push to talk button to speak. Thus the meeting can continue with the button released, then they can pause the meeting and push the button and speak more... Something like that is my understanding. Currently I have one of the new Ubiquity phones on my desk. There handsets have a mute button, or if you want a speak button, but a phone running under Android for government usage might leave some questions unanswered. If your phones have some functions keys, you'd have a look at the MuteAudio function and map the states to DTMF sequences, which in turn are mapped to the function keys. This makes you rather independent from any hardware and you might adapt the behavior depending on what your clients wishes will finally be, if they ever find out themselves. What I don't understand is why the normal mute button on most headsets is not sufficient. jg -- _ -- 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] PTT push to talk solution
I am looking for a push to talk solution does anyone know of a good PTT phone one that works with asterisk. I'm not talking about polycom fake PPT... I'm talking about a real call into Asterisk and having to push a button on a headset or the phone to actually talk. not multicast talk like polycom. I wish polycom had a real PTT headset but I cannot find one, I like their phones. Cisco has a PTT headset but seems only for 7960 model. Those phones are older and diffucult time find a new one and hard to get SIP on 7960. So is there a PTT phone out there that works great with asterisk ? I am not sure whether I really understood your question. It looks to me that the PTT functionality can easily be achieved using the mute button that most phones and headsets have. One could even implement it independent of any specific phone using the Asterisk function MuteAudio(). One could use the DTMF features for signaling. jg -- _ -- 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] Modifying CDR values from a hangup extension in Asterisk 13
I'm trying to migrate from Asterisk 1.8 to Asterisk 13 and can't figure this one out. I'm
pretty sure the question has been already asked, but I failed to find a solution.
Can you modify CDR values in an h-extension?
My cdr.conf contains:
[general]
enable=yes
unanswered=yes
endbeforehexten=yes
initiatedseconds=no
batch=no
The diaplan contains a simple h extension
exten = h,1,NoOp(${CDR(userfield)})
exten = h,n,Set(CDR(userfield)=changed)
exten = h,n,NoOp(${CDR(userfield)})
In the same context I execute:
exten = 10,1,Set(CDR(userfield)=empty)
exten = 10,n,Dial(SIP/10)
The h extension outputs two lines with userfield set to empty. I would expect the second
one to be changed. It seems that I can read the CDR values, but I can't change them. Is it a
bug or a design thing? Am I missing something?
I am not working with h-extensions myself, but the docs
(https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_cdr) say something like this:
|endbeforehexten|
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_cdr#Asterisk13Configuration_cdr-general_endbeforehexten
|Boolean|
|1|
|false|
Don't produce CDRs while executing hangup logic
This would indicate that at least writing is disabled.
jg
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Re: [asterisk-users] showing sip number insted of pri number
Thank you for your reply. Can you please guide me how to spoof the number in outbound call. To number in my chose. I don't know any telco or national law that would allow setting arbitrary numbers. You might have been assigned a certain number rage, and you could pick any from that range. If you set a number that does not belong to your range, the telco will typically substitute it with a standard number. Let's say you've got 100 numbers, e.g. 1234-0 to 1234-99, then anything out of that range, e.g. , is likely to show up as 1234-0 on the callee's phone. Having said that you also need to coordinate your efforts with your telco. You need to check several transmitting and switching facilities, like CLIP, CLIR, COLP, COLR, possibly CNIP. CLIP and COLP comes with different flavors. I'd say that the details are outside of what can be handled here. jg -- _ -- 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 through the GUI 35 cisco ip phones -spa502g
I am new to Asterisk and VOIP so I am trying to find a decent howto guide on setting up cisco ip spa502g VOIP phones. I have found a interesting document on Cisco website but unable to access it. https://supportforums.cisco.com/document/37376/asterisk-configuring-cisco-spa5xx-phones-web-ui -- I have contacted them for access and waiting on their reply. Can someone please suggest some other guides that will assist me. I am using a couple of older Sipura/Linksys/Cisco SPA phones myself. Some features maybe lacking, but there are no special setup procedures. http://spakonfig.de/ shows typical configurations, where special settings are marked with a red color. You should stay away from the regional parameters unless you know what you are doing. You need to pay attention to the Dial Plan. The downside of spakonfig.de is that it is in German, but it might still be helpful. jg -- _ -- 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] Windows Asterisk Help
Downloaded latest version of Asterisk from www.asteriskwin32.com and installed on Windows 7. Here is my sip.conf [general] context = demo ; Default context for incoming calls bindport = 5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ; Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =16194077214:password@69.59.234.67:5060/202 [authentication] [3000] type = friend context = default username = 3000 host = dynamic mailbox = 3000 dtmfmode = rfc2833 [3001] type = friend context = default username = 3001 host = dynamic mailbox = 3001 dtmfmode = rfc2833 [3002] type = friend username = 3002 context = default host = dynamic mailbox = 3002 dtmfmode = rfc2833 [vonage-out] username=16194077214 type=friend secret=password port=5061 nat=yes host=69.59.234.67 fromuser=16194077214 fromdomain=69.59.234.67 dtmfmode=rfc2833 auth=md5 [vonage202] username=16194077214 ;type=friend type=peer ;type=user secret=password port=5061 nat=yes insecure=port,invite host=69.59.234.67 fromuser=16194077214 fromdomain=69.59.234.67 ;dtmfmode=inband context=from-pstn canreinvite=no ;auth=md5 disallow=all allow=ulaw ;allow=alaw ;allow=g729 ;allow=g723 Here is my extensions.conf [from-pstn] ;exten = 16194077214,1,verbose(0, hello) exten = 16194077214,1,Answer; exten = 16194077214,n,SayUnixTime() exten = 16194077214,n,Hangup I am able to connect with Asterisk on the first try after fresh load, but not on the subsequent tries. I have to re-reload sip.conf and extensions.conf to connect with Asterisk. Looking at the logs, it seems like a registration issue. So I set minexpirty and maxexpirty that seems to have no effect. can post the logs, if someone wants me to. Your kind help is appreciated. Best regards murthy www.asteriskwin32.com hosts only a very very old version of Asterisk (1.2.something). What speaks against setting up a small virtual machine to host a recent version of Asterisk? jg -- _ -- 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] Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?
Hmm ok - I made sure to run make config in the Asterisk source folder which installed the correct scripts into rc.d and so forth. I then did chkconfig asterisk on and rebooted the box. The parameters remain the same, asterisk is there if you do a ps -aux | grep asterisk but it still is in a non-working state and not contacible via asterisk -r. Since it is an old box and the reason for trying to get it going is mostly academic, I think I'm just going to dump the box and reformat it with Centos 7. Strange though, I have installed about 17 other boxes exactly this way on broadly the same hardware and all are currently running fine with Centos 6.5 and Asterisk 1.8.11.0 Thanks anyway, the problem is clearly deeper than I though since even the official way you detail above fails to start Asterisk as an account that can start it on system boot. Even when root is the only account on the machine - which leads me to believe I have some basic error in my Centos 6.5 installation so I'll just try it again or try Centos 7. Okey-dokey. What happens when you start asterisk with asterisk -c from a root account? jg -- _ -- 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] Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?
I'm trying to get Asterisk 1.8.11.0 to start automatically when my Centos 6.5 box boots. I've done this many times before, but for some reason, on this box and hardware (older Core i3 system, 4GB RAM) I cannot get Asterisk to be contactible after boot. E. g. in rc.local I have, as the last line --- asterisk --- as in all my other Asterisk boxes with Centos 6.5 and Asterisk 1. 8.11.0 This -does- start asterisk on boot, but you cannot connect to it using asterisk -r the error being ... Depending on the hardware you are using, simply calling asterisk might not be enough, as there could be dependencies on third party drivers. Depending on how asterisk was installed, one probably also has to look at various permissions. For example, asterisk -r might fail simply because you are calling it from an account with insufficient rights. It's difficult to tell given your information. Just recently, I was caught by a user inflicted problem and spent some time evaluating SIP messages... Maybe the following will help. If you look into the contrib/init.d directory (inside the src tree) you'll find the rc.redhat.asterisk script. Rename it and put it into the /etc/init.d directory and issue chkconfig --add asterisk as well as chkconfig asterisk on and your problem should be solved. You can check the current settings with chkconfig --list asterisk. The Redhat script works nicely under CentOS. jg -- _ -- 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] Always 486 Busy Here for anonymous calls
I am running an Asterisk PBX 11.6-cert10 with about 20 SIP phones and recently one of the phones (Snom 720) always returns 486 Busy Here when calling anonymously. It's only a single phone, the rest works as expected. I checked the phone's settings and there are no differences in the configuration compared to the rest. I do not expect that Asterisk is the problem, but does someone know under which circumstances this kind of problem can occur? This is just a feedback that the problem is now solved. A user had the great idea to add anonymous@anonymous.invalid to the local deny list. Very funny. jg -- _ -- 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] Always 486 Busy Here for anonymous calls
Hi! I am running an Asterisk PBX 11.6-cert10 with about 20 SIP phones and recently one of the phones (Snom 720) always returns 486 Busy Here when calling anonymously. It's only a single phone, the rest works as expected. I checked the phone's settings and there are no differences in the configuration compared to the rest. I do not expect that Asterisk is the problem, but does someone know under which circumstances this kind of problem can occur? jg -- _ -- 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 no voice
I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-01d1 setting write format to g729 from alaw native formats 0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: channel.c:5254 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type g729, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x8 (alaw)/0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x8 (alaw)/0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-01d1 setting write format to g729 from alaw native formats 0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: channel.c:5254 set_format: Unable to find a codec translation path from 0x8 (alaw) to 0x100 (g729) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type g729, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) [Jul 15 18:59:55] WARNING[8752]: chan_sip.c:6773 sip_write: Asked to transmit frame type alaw, while native formats is 0x100 (g729) read/write = 0x8 (alaw)/0x8 (alaw) In my sip.conf I have: disallow=all allow=alaw allow=ulaw allow=ilbc allow=g729 allow=g723 allow=gsm I tried with allow=all, too, but it results in no communication on all numbers... Could someone help me? How is the 4th phone configured? You could also enable SIP debugging to get more information about the problem. jg -- _ -- 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] Sending E-Mail from voicemail
Hi Gods! I need a change on my Voicemail configuration, but I can't experiment now, since the system is in use... Your fellow compatriot Immanuel Kant would say sapere aude (dare to know). As I said, I can't just try... Well, https://www.youtube.com/watch?v=yTCDVfMz15M and https://www.youtube.com/watch?v=1VGkmPF1CNo Have you ever thought of setting up a virtual machine to (e.g. VirtualBox) for testing and developing? Most phones allow several SIP accounts, so you could test this with your existing equipment. jg -- _ -- 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] CDR in an MySQL-Database
Hi list!
I'd like to save all information about calls (CDR) in a MySQL-Database.
I created the DB and a user for Asterisk on a separate server, then I
configured my cdr_mysql.conf so:
[global]
hostname=192.168.10.3
dbname=asterisk
table=cdr
password=MYSECRET
user=asterisk
port=3306
and my cdr.conf so:
[general]
enable=yes
unanswered = yes
safeshutdown=yes
[mysql]
usegmtime=no
loguniqueid=yes
loguserfield=yes
accountlogs=yes
I created the table in the DB so:
CREATE TABLE IF NOT EXISTS `cdr` (
`id` int(11) unsigned NOT NULL AUTO_INCREMENT,
`calldate` datetime NOT NULL DEFAULT '-00-00 00:00:00',
`clid` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`src` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dst` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`dcontext` varchar(80) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastapp` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`lastdata` varchar(200) COLLATE utf8_bin NOT NULL DEFAULT '',
`duration` float unsigned DEFAULT NULL,
`billsec` float unsigned DEFAULT NULL,
`disposition` enum('ANSWERED','BUSY','FAILED','NO ANSWER','CONGESTION')
COLLATE utf8_bin DEFAULT NULL, `channel` varchar(50) COLLATE utf8_bin DEFAULT
NULL, `dstchannel` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`amaflags` varchar(50) COLLATE utf8_bin DEFAULT NULL,
`accountcode` varchar(20) COLLATE utf8_bin DEFAULT NULL,
`uniqueid` varchar(32) COLLATE utf8_bin NOT NULL DEFAULT '',
`userfield` float unsigned DEFAULT NULL,
`answer` datetime NOT NULL,
`end` datetime NOT NULL,
PRIMARY KEY (`id`),
KEY `calldate` (`calldate`),
KEY `dst` (`dst`),
KEY `src` (`src`),
KEY `dcontext` (`dcontext`),
KEY `clid` (`clid`)
) ENGINE=InnoDB DEFAULT CHARSET=utf8 COLLATE=utf8_bin AUTO_INCREMENT=1 ;
Then I restarted Asterisk (core restart now).
Unfortunately it does not work, since I get on boot:
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1468 load_mysql_config:
MySQL RealTime: No database user found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1474 load_mysql_config:
MySQL RealTime: No database password found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1480 load_mysql_config:
MySQL RealTime: No database host found, using localhost via socket.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1486 load_mysql_config:
MySQL RealTime: No database name found, using 'asterisk' as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1492 load_mysql_config:
MySQL RealTime: No database port found, using 3306 as default.
[Jul 6 21:20:39] WARNING[9735]: res_config_mysql.c:1509 load_mysql_config:
MySQL RealTime: No database socket found (and unable to detect a suitable path).
And of course:
OpenWrt*CLI cdr show status
Call Detail Record (CDR) settings
--
Logging:Enabled
Mode: Simple
Log unanswered calls: Yes
* Registered Backends
---
cdr-custom
Asterisk 1.8 runs on an OpenWRT-Switch.
Any idea?
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
Did you study this: http://www.asteriskdocs.org/ ?
jg
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Re: [asterisk-users] Custom header when busy
Thanks for the tip. Our goal is to know that call-limit is triggered. And later analyze this info, maybe do some action. Yes, we can parse CDRs or execute AGI script but we do not want inmplement this logic on Asterisk because it can affect performance. 02.07.2015, 15:31, jg webaccounts...@jgoettgens.de: Is there any chance to create feature request for that useful functionality? 02.07.2015, 14:03, Rusty Newton rnew...@digium.com mailto:rnew...@digium.com: On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru mailto:r...@yandex.ru wrote: Hi, all Is there someway ability to insert custom Header to SIP 486 message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment. I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently. I think that Asterisk cannot handle this in general. There might be further call-limit restrictions coming from the individual settings of your phones. I think the easiest way for inhouse calls is to use Action URLs (if supported by the phone) and setup a a finite state machine externally to handle your needs. CDRs also work, but you have to do a lot more because you need to evaluate the time information for ringing, talking, simultaneous calls, etc. A small state machine is easier to handle. I do this kind of stuff when I have to install new boxes to get an overview of various statistics. jg -- _ -- 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] Custom header when busy
Is there any chance to create feature request for that useful functionality? 02.07.2015, 14:03, Rusty Newton rnew...@digium.com: On Wed, Jul 1, 2015 at 4:46 AM, r...@yandex.ru mailto:r...@yandex.ru wrote: Hi, all Is there someway ability to insert custom Header to SIP 486 message, when HANGUP application is invoked? Our use case is to set that Header, when call-limit is reached, to analyze elsewhere, but we do not want to set some custom causecode in HANGUP application because this can confuse a calling equipment. I only know of the SIPAddHeader application which lets you add headers when used before Dial, so I don't think you can do this currently. I think that Asterisk cannot handle this in general. There might be further call-limit restrictions coming from the individual settings of your phones. I think the easiest way for inhouse calls is to use Action URLs (if supported by the phone) and setup a a finite state machine externally to handle your needs. jg -- _ -- 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 email to fax
being honest with i have been lost on what to do. all i want is sent from my email a pdf file and then the server will sent it as fax. what settings do i have to do regarding emailing to the server? what other settings do i have to do? is there a guide on that? *Sent:* Friday, June 26, 2015 at 7:28 PM *From:* Tiago Geada tiago.ge...@gmail.com *To:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] asterisk email to fax we use a PHP web page, that takes a few formats, PDF being the most common, anc convert it to TIFF. If conversion succeeds we allow to download the TIFF file as a preview. Then the user confirms and the PHP places a .call file in asterisk spool On 25 June 2015 at 19:51, Ryan, Travis ry...@oscarwinski.com wrote: I hope his mother in law doesn’t live with him. That’s a support issue for sure. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Kevin Larsen *Sent:* Thursday, June 25, 2015 2:50 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] asterisk email to fax Since the O.P. said he's using it for his home office, I think he'll be able to control user expectations :-) I provide tech support to my parents on all their computers. The amount of annoyance I have dealt with in the last few months over the fact that a recipe program and various card making programs designed for Windows 3.1/95 won't run on my mom's Windows 7 64 bit computer tells me you are not as right as you think you are. This is not a question of settings. You must decide yourself what to do and there are various options. Asterisk is only responsible for the transport. You need to look at the following tasks: - get the file or files into the asterisk box - convert to a faxable tiff format - generate a call file and put the tiff file where it belongs (but there are other methods) - get fax report Of course, at first you must configure Asterisk to accept facsimiles, which depends on which technologies you need. But everything is nicely documented. jg -- _ -- 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] no samples for gsmtolin
If I call a number from the phone of my wife, I get this warning: [Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for gsmtolin I think this is related to silence suppression. Either ignore it, or find the device that does this and disable silent suppression. jg -- _ -- 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 again I'd like to configured my Asterisk to use german sounds for the Say-commands... Generate your own German sound files, it's not difficult, but rather time consuming. A couple of years ago I suggested to donate my own files, but the problem were the license conditions of the text to speech software. If there is enough interest, I could contact the vendor again and ask about acceptable compensation. Of course, you and a few more must be willing to pay for the sound files. The sound quality of old gsm Amooma files is pretty bad, but I don't know from where you can get them and the old Pforzheim files are incomplete. jg -- _ -- 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
I think there are many german users in this ML, that use Asterisk with the new line of Deutsche Telekom (Magenta Zuhause). My ISDN will be converted in Juli (Kaboom-day at Juli, the 3rd...) and right now I can just hope, that I configured my Asterisk well to work with Deutsche Telekom, but I cannot be sure, since I can't test it... So my question: can someone using Asterisk with Deutsche Telekom contact me (PN), so that we can compare the sip.conf? It doesn't really depend on your sip.conf and Asterisk. Your gateway/router will be the major problem. My summer project will be to look at session border controllers which separate the local PBX features from the WAN side. The most simple setup would be to use a FritzBox, which I think can handle SIP trunks as well. I'll try my luck with pfSense, where I'll have a look at Asterisk, FreeSwitch and possibly Kamailio for the telephony part. You need only basic switching capabilities here and the focus is on security (geo stuff, access patterns, ...) and access rights, possibly patching some SIP headers. It's also nice to have an intrusion detection system like Snort and a defined interface for packet capturing. I have already a lot of experience with pfSense and I appreciate all the security, monitoring features and stability, but I don't really know whether in fall I'll have something that can be recommended. Anyway, my attempts to setup reliable SIP and IAX2 protocols so far failed. I always found workarounds, but in case of interruptions, DSL line termination, IP changes etc, I frequently ended up with wrong intermediate ports and failing connections with routers that I don't know. For me it is easier to have everything with pfSense under my own control instead of figuring out the peculiarities of certain NAT implementations. The worst thing that can happen to me is a customer with a 15 year old router I've never seen before. It is usually easier to say that a special router/SBC is part of the deal instead of guessing how much hassle it could be this time. If I look at the complexity of my routers' packet filtering, it makes definitely sense to separate gateway from internal functionality. One could say that cascaded Back-to-Back-User-Agents look peculiar, but once you start to think about maintenance, it makes a lot of sense (to me and momentarily). jg -- _ -- 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
It doesn't really depend on your sip.conf and Asterisk. Your gateway/router will be the major problem. My summer project will be to look at session Are you sure? Right now I'm using an italian SIP-Provider (Messagenet), configured in my sip.conf and I can receive calls without any problem... So, I don't think, I have to expect problem on my NAT (anymore... initially I had some problems...). There's nothing special, only if you want to set up your own infrastructure for finer control. jg -- _ -- 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] Curious problem with NAT
Very strange is, that I have a very poorly audio-quality, if I use my cellphone in my WLAN and connect to my Asterisk. With THE SAME USER, but from a PC in the same Network, the audio quality is perfect. Any idea? Did you check which codecs are active? What does sip show channelstats say? jg -- _ -- 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] Signaling incoming call
Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with fooling... My phone will receive calls from 3 numbers. All that was done in my dialplan. Now, it would be nice, if I can signaling on the phone which number will be called, so that, for example, if I receive a call for +4935 I get a message on the display or the phone ring with a particular tone, and if I receive a call for +49351222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. I don't know your phones, but there are multiple ways to achieve that. By far the easiest method is to work with multiple SIP identities. You can adjust quite a few parameters, like display text, ring tone, timings, forwarding While you are busy with this, you can add additional accounts that operate as intercoms (baby monitors) so you don't have to wait for an answer. Interesting exercise, but might disturb peace in the house. If your phone supports only a single identity, then you have to adjust caller ids, etc with Asterisk. jg -- _ -- 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] Debugging dialplan
Since I think, I have a problem in my dialplan, how can I debug it? It would be very useful a command in Asterisk CLI to ask Asterisk what it would do if the number X call the number Y. Something like exim -bt, if someone here know the SMTP-daemon Exim... Is there such an option in Asterisk? Yes, it is called core set verbose 42, the other options is core set debug 42. Enjoy the show! Once you are more familiar with *, you might want to have a look what you can do with logger.conf. jg -- _ -- 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] Seeking advice about ISDN BRI Cards
Thank you all for valuable input, another question: when do I actually need the echo cancellation (hardware / on board /on module ) ? It depends on your environment. If there are still analog devices in addition to VoIP, I'd say always, but Asterisk has a rudimentary echo canceller already on board. The Telcos use echo cancellers themselves, but it cannot hurt to have a hardware canceller on your BRI card. Nowadays I see more problems with reverberation in connection with cheap speakerphones or simple mics and speakers on PCs, but that's a different story. jh -- _ -- 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 make asterisk work with remote mysql database?
I am trying to make asterisk 11.7 work on Ubuntu 12 (amd64). I would like asterisk to not use the inbuilt sqlite database. Instead, I want it use a remote mysql database. Is this possible ? If yes, is there any good HowTo on this ? Running 'make install' installed my asterisk successfully, but no conf files were copied to /etc/asterisk. Nor was a service start script placed in /etc/init.d. Is there normal ? fixable? One additional problem is that, after installation, the CLI has no odbc command. Why could this be ? I suspect that for asterisk to work with remote mysql database, odbc functionality needs to be fully workable. Installed from sources or from packages? If you install from sources you must make sure that the ODBC parts, dev stuff as well, must be installed. When you add something to your system, you also need to run ./configure again. Then try make menuselect to check your configuration. If you cannot select an item, there are usually hints on what the resource depends. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Am 27.04.2015 um 14:53 schrieb akhilesh chand: Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior helvio.lis...@gmail.com mailto:helvio.lis...@gmail.com wrote: Hi Akhilesh, SIP protocol use port 5060 (default) and many other ports to stablish calls. You need to check if there is AWS firewall rule that allow your communication from your client external IP and your AWS host. Also, think in use IAX intead of SIP, because SIP protocol has many trouble when used with NAT, also IAX protocol use only one port (4569) to everything. When i need allow external clients (throught NAT or not) i used to use IAX. If you want i can help you in your environment (SIP or IAX). Att, Hélvio Junior SafeId - Gestão de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br http://single-sign-on.com.br helvio.jun...@safetrend.com.br mailto:helvio.jun...@safetrend.com.br On 24/04/2015 06:35, akhilesh chand wrote: Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh -- _ -- 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 Did you read this: http://www.asteriskdocs.org/ ? Having said that, you might still run into some NAT-related problems, if you use a normal router. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error writing CDR
Hi All I have dozens of these messages on CLI complaining about database connection and error writing CDR to disk. The curious thing is I can find them all inside the database. I selected them using uniqueid and manually compared each column with the cdr_adaptive_odbc.c error line. mysqlcheck -a -e -v DBase and mysqlcheck -c -e -v DBase both returned OK for all tables. Environment is: in production Asterisk 11.7.0~dfsg-1ubuntu1 Ubuntu 14.04.1 LTS Any thoughts? Thanx Ethy [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) Hi Ethy, why date and time are empty? At least date is used as a unique key and a unique key has to be unique. In other words, the same key can not exist twice like in your case. Check why there is no date and time anymore ... Or define your table with and independent primary key that gets added automatically: mysql describe cdr; +--+--+--+-+-++ | Field| Type | Null | Key | Default | Extra | +--+--+--+-+-++ *| id | int(11) | NO | PRI | NULL| auto_increment |* | clid | varchar(80) | NO | | | | | src | varchar(80) | NO | MUL | | | | dst | varchar(80) | NO | | | | ... | lastapp | varchar(80) | NO | | | | | lastdata | varchar(80) | NO | | | | | duration | int(11) | NO | | 0 | | | billsec | int(11) | NO | | 0 | | | disposition | varchar(45) | NO | | | | | start| datetime | NO | MUL | -00-00 00:00:00 | | | answer | datetime | NO | | -00-00 00:00:00 | | | end | datetime | NO | | -00-00 00:00:00 | | | uniqueid | varchar(45) | NO | | | | ... Just in case you get bogus records with offending primary keys due to some other problem, you would still have valid data base entries and you would be able to look at the pattern. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error writing CDR
Hi All I have dozens of these messages on CLI complaining about database connection and error writing CDR to disk. The curious thing is I can find them all inside the database. I selected them using uniqueid and manually compared each column with the cdr_adaptive_odbc.c error line. mysqlcheck -a -e -v DBase and mysqlcheck -c -e -v DBase both returned OK for all tables. Environment is: in production Asterisk 11.7.0~dfsg-1ubuntu1 Ubuntu 14.04.1 LTS Any thoughts? Thanx Ethy [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to MyAsterisk-asterisk [MyAsterisk-asterisk]... [Apr 25 10:56:56] WARNING[19013][C-02cb]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1527 odbc_obj_connect: Connecting MyAsterisk-asterisk [Apr 25 10:57:01] NOTICE[19013][C-02cb]: res_odbc.c:1559 odbc_obj_connect: res_odbc: Connected to MyAsterisk-asterisk [MyAsterisk-asterisk] [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:645 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 23000: [MySQL][ODBC 5.1 Driver][mysqld-5.5.40-0ubuntu0.14.04.1-log]Duplicate entry '-00-00 00:00:00-1234-CLIENT_ID' for key 'PRIMARY' (133) [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:657 ast_odbc_prepare_and_execute: SQL Execute error -1! Verifying connection to MyAsterisk-asterisk [MyAsterisk-asterisk]... [Apr 25 10:57:01] WARNING[19013][C-02cb]: res_odbc.c:761 ast_odbc_sanity_check: Connection is down attempting to reconnect... [Apr 25 10:57:02] WARNING[7666]: chan_sip.c:4409 __sip_autodestruct: Autodestruct on dialog '34f3f3481b8d1e4772dc111f42d13...@ip.ip.ip.ip:5060' with owner SIP/CLIENT_ID_1-0178 in place (Method: BYE). Rescheduling destruction for 1 ms [Apr 25 10:57:06] NOTICE[19013][C-02cb]: res_odbc.c:1527 odbc_obj_connect: Connecting MyAsterisk-asterisk [Apr 25 10:57:06] NOTICE[19013][C-02cb]: res_odbc.c:1559 odbc_obj_connect: res_odbc: Connected to MyAsterisk-asterisk [MyAsterisk-asterisk] [Apr 25 10:57:06] WARNING[19013][C-02cb]: cdr_adaptive_odbc.c:739 odbc_log: cdr_adaptive_odbc: Insert failed on 'MyAsterisk-asterisk:cdr'. CDR failed: INSERT INTO cdr (dst,accountcode,clid,src,dcontext,channel,dstchannel,lastapp,duration,billsec,disposition,amaflags,userfield,lastdata,uniqueid) VALUES (blahblahblah, ... ,'1429970147.612') Can you post the output of describe schema;? At least the first error message is probably related to a not so optimal primary key definition. jg -- _ -- 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] Grandstream GXP2140
I have a customer looking to deploy about 20 Grandstream GXP2140 phones. Normally they would deploy Yealink brand phones but they want a phone with gigabit pass through and the Yealinks with gigabit are too expensive for their budget. Does anyone on the list have experience with the GXP2140? Is it a reliable phone? Does anyone have recommendations for other phones with gigabit pass through? I'd be generally careful with the second ethernet connection. One should look at the chipset of the phone. I had pretty bad experiences with somewhat older TI based phones, regardless of the manufacturer. The problems became apparent in mixed environments, where some connections were gigabit and others not. It can be a nightmare, if you have to offer support. The best bet is to buy one, and check the performance of the connections. I use some GrandStream products myself and the product quality is now much better compared to a couple of years ago. jg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to install asterisk in AWS cloud
I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 Just guessing. Did you call ./configure? jg -- _ -- 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] Script to Program Snom Phones
Does anyone know how to program Snom phones using a Mac addresses like what is done with the Ciscos. I have about 50 extensions to be programmed and I am hoping there is a way on Asterisk to program extensions on the snom phones. Please assist. What do you mean with 50 extensions? Snom phones allow to define a directory, where you can export and import a simple text file. There might also be a way to automate this using one of the provisioning methods. jg -- _ -- 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] 4 Port PRI
I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes ;echocancelwhenbridged=no ;rxgain=0 ;txgain=0 callerid=asreceived musiconhold=default group=1 overlapdial=yes signalling=pri_cpe context=extensions channel = 1-15,17-31 jbenable= yes jbforce= yes jbmaxsize= 120 jbimpl= fixed jbresyncthreshold= 1000 PRI or BRI? Which card are you using? Typically the installation script or procedure lets you configure each span. You seem to have 4 spans for either 8 or 128 (EuroISDN) channels. jg -- _ -- 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] Disabling Ringing/Alerting
Can anyone please guide us if there's any way of disabling alerting/ringing in asterisk when a call is placed to any subscriber. What we want is the channel establishment as it happens during a call progress but the subscriber should not ring. Is this possible in asterisk? I think this is not an Asterisk feature. It's up to the phone to decide what to do with an invitation. There are typically multiple configuration options to take care of questions like yours. jg -- _ -- 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] Yealink t26 and T28 Panels
Hi! We have a strange a strange issue at a client they have 3 panels on their phone and every so often the panels reboot themselves yet the phone stays on. We decided to replace the T26 for a T28 to see if it fixes the issue and still have the exact same issue. Has anyone seen this before? I frequently use the newer T48G and T46G phones with the EXP40 expansion module. There are issues, if you are logged into the phone via the webinterface as an admin. Among other things, the display is not properly updated and wrong numbers may get dialed. Some time ago, there was a firmware update and I am not aware of any stability issues at the moment. How do you supply power? 3 expansion modules + the phone and a cheap POE switch could be critical. It may not be the power itself, but the correct handling of energy saving states. jg -- _ -- 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] which libsrtp ?
I've been having some issues with srtp. so I checked which version of libsrtp I built asterisk 11.6 against. I'm on fedora 21, so libsrtp-1.4.4-13.20101004cvs.fc21.x86_64. From https://github.com/cisco/libsrtp it seems that latest release is 1.5.1, released a couple of weeks ago. I'm not a fan of the bleeding edge, but using a version 4+ years old seems strange even to me. But, on the other hand, it seems to Work For Me. Anybody using 1.5.1 ? It shouldn't matter, provided it is the current version 1.4.2. On RedHat like systems I use the srtp epel package, but I also work with yum-priorities. So far I have not seen any difficulties. Building from source is also very easy. jg -- _ -- 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] TLS, SRTP, Asterisk11 and Snom870s
Am 03.03.2015 um 18:16 schrieb James B. Byrne: CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten this set-up (Asterisk11 with Snom870s using TLS) to work and if so could you provide the details? I have this in Asterisk sip.conf (loaded through FreePBXs sip_general_additional.conf). tcpenable=yes tlsenable=yes tlscertfile=/etc/pki/asterisk/ca.harte-lyne.hamilton.asterisk.crt tlscafile=/etc/pki/tls/certs/ca-bundle.crt tlsdontverifyserver=yes tlscipher=ALL tlsclientmethod=tlsv1 And I have this for the test device context: [41712] deny=0.0.0.0/0.0.0.0 secret=NearlyANastyThat dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=no port=5060 qualify=yes qualifyfreq=60 transport=tls,udp,tcp avpf=no force_avp=no icesupport=no encryption=yes callgroup= pickupgroup= dial=SIP/41712 mailbox=41712@device permit=192.168.6.0/255.255.255.0 callerid=James B Byrne 41712 callcounter=yes faxdetect=no cc_monitor_policy=generic If I change the transport setting to TLS then I get this reported: [2015-03-03 11:10:08] ERROR[22244]: tcptls.c:875 ast_tcptls_client_start: Unable to connect SIP socket to 192.168.6.112:5060: Connection refused I cannot seem to configure the Snom870 to listen for TCP on 5060. There is a setting for that on the phone but it seems to have no effect (it always returns to NO following a reboot). The Snom website says that the option is not available in FW8.5 and later. It does not inform one of whether that the phone listens by default or not on FW8.5+, only that the option has no effect. It also does not say, as far as I can find, whether Snom870s listen for TCP at all or on what port. One may infer that since these devices purport to support TLS that the answer is yes and that TCP5061 is a likely candidate. But they do not seem to come right out and say so anywhere. In a section devoted to the Snom370, which is a model that we do not employ, there is reference to DNS SRV RRs. The inference drawn from the examples given is that these will control what ports the Snom will listen on for which services. We have such records in our DNS zone. They look like this: ;# Configure sip/sips service records (VOIP) ;HOST TTL CLASS TYPEORDER PREF FLAGS SERVICE REGEXP REPLACEMENT 300 IN NAPTR 50 50 s SIPS+D2T_sips._tcp.harte-lyne.ca. 300 IN NAPTR 90 50 s SIP+D2T _sip._tcp.harte-lyne.ca. 300 IN NAPTR 100 50 s SIP+D2U _sip._udp.harte-lyne.ca. ;HOST TTL CLASS TYPEORDER PREF PORTTARGET _sips._tcp.harte-lyne.ca. 300 IN SRV 10 10 5061voinet09.hamilton.harte-lyne.ca. _sip._tcp.harte-lyne.ca.300 IN SRV 10 10 5060voinet09.hamilton.harte-lyne.ca. _sip._udp.harte-lyne.ca.300 IN SRV 10 10 5060voinet09.hamilton.harte-lyne.ca. However, our phones are configured to use SIP accounts having the form account@ipv4-addr. I doubt greatly that the Snom870s will perform a reverse DNS lookup on the provider's IPv4 to discover the forward zone domain and thus I do not believe that SRV RRs can help us in this instance. They certainly do not seem to have any effect. Asterisk seems not to distinguish between 5060 and 5061 regarless of protocol. I am not sure then how to proceed. Is there a way to force Asterisk to talk to port TCP5061 on a specific device? Is this an exclusive setting? This long background is by way of asking for help. If I have not provided specific information that is significant to this problem then I will do so if asked. What I am attempting has to be possible. Somehow. And somebody must have already accomplished this. Somewhere. Forget about the reverse DNS stuff for the moment. Do simple SIP accounts (without SRTP/SRTP and deny/permit stuff) work? Enable SRTP, but you likely need the AES-80 fro SRTP Auth-tag. Then try the rest. jg -- _ -- 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] IAX port
I get an occasional similar problem, we have Mikrotik firewalls and from tcpdump monitoring on the asterisk boxes I can see that the firewall (unbidden) has changed the IAX port. Usually a firewall reset and sometimes PBX reset combination fixes it. Its odd as its only one direction, occurs rarely and with no obvious driver. So IAX is happy in one direction but not the other. And I can see packets in the unhappy point arriving on the wrong port. I couldn't fix it without kicking the router/firewall so I would say its a router problem in the Destination NAT process. Cheers Duncan Port is changed when NAT is applied from LAN to WAN. While UDP session is maintained as ESTABLISHED, that port should not change. If your peer changes constantly of session port could be UDP session is too short in NAT table on routers. You can try setting qualify=1000 (which is in ms. Default is 2000), and see if peer keeps same port. Regards. voip-info.org also has an entry about general NAT related issues, which could be relevant here I do not seem to have problems with Netgear firewalls, but other firewalls show this effect. So far it happened only on a single side, such that calls work from the other side. I already checked the open ports with nc/ncat/netcat as UDP sender and receiver on the other end. The ports are open, even when the arbitrary ports are used by Asterisk. I'll need to read a bit more and evaluate my pcap traces and possibly ask the router vendors. Thank you for your efforts. jg -- _ -- 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] IAX port
UDP timeout being too short is another thing I've experience with firewalls (admittedly limited and once removed experience). Actually, this one can be a (mild) problem on Draytek routers and can be resolved by telnetting into the router and using the portmaptime command. Also, turn of stateful packet inspection if it is an option. In this case it is a TP-Link VPN router (TL-ER6...). -- _ -- 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] IAX port
Some firewalls have a 'consistent NAT' option that needs to be enabled, otherwise you get the symptoms described. While reading about NAT, I came across this web site: http://nattest.net.in.tum.de/ The test tool looks at various NAT related properties and prints the results related to TCP/UDP binding properties, TCP/UDP hole punching, etc. In my case a very short value was reported for the UDP timeout, such that depending on the sequence of packets, the entry in the mapping table might already have been deleted. This could explain the random nature of my connection problem. Port predictability does not seem to be a problem. Does that make any sense? jg -- _ -- 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] IAX port
Hi! Sometimes IAX peers are not reachable and with iax2 set debug on I get something like this Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 1 DCall: 01200 79.233.155.174:49153 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 01200 DCall: 1 79.233.155.174:49153 I am not sure what causes port 4569 to be replaced an an arbitrary port, which could be the reason for my problem. Does someone know whether this is a router related problem? jg -- _ -- 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] IAX2 problem for WAN connections
Hi, I am trying to connect two Asterisk servers using IAX2. Everything works fine when I couple them within a LAN segment, but not when I connect them using WAN connections. I made sure that the routers' ports are mapped properly and checked this with additional ssh rules. ServerA is a Raspberry box with the vendor's Asterisk version 1.8.13.1 and ServerB is normal CentOS 7 box with Asterisk 13.1. Calling from ServerB to ServerA works, but not vice versa. The only odd thing that appears to me is the different perceived port on ServerA. ServerA*CLI iax2 show registry Host dnsmgr Username PerceivedRefresh State 80.152.xxx.xxx:4569 N ServerA 79.233.yyy.yyy:45697 60 Registered ServerB*CLI iax2 show registry Host dnsmgr Username Perceived Refresh State 79.233.yyy.yyy:4569 N ServerB 79.233.yyy.yyy:4569 60 Request Sent Does someone have an idea at what to look in detail? jg -- _ -- 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] IAX2 problem for WAN connections
I found a way that works. Essentially, I deleted the register lines and added the hosts with deny all and specific permit specs. I don't know why it works, but it does. jg -- _ -- 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] IAX2 problem for WAN connections
On Thursday 05 Feb 2015, jg wrote: Calling from ServerB to ServerA works, but not vice versa. The only odd thing that appears to me is the different perceived port on ServerA. Does someone have an idea at what to look in detail? Look in /etc/asterisk/iax.conf in the first instance. Basically I used the example from the Asterisk book Connecting Two Asterisk Boxes Together via IAX and there is not a lot to see: ; Server A [general] ; this boxremote IP register = ServerA:very_sec...@80.152.xxx.xxx disallow=all allow=alaw allow=ulaw allow=gsm jitterbuffer=no forcejitterbuffer=no autokill=yes ; the other box [ServerB] type=friend trunk=no auth=md5 encryption=yes secret=very_secret context=from-ServerB qualify=yes host=dynamic ; end of Server A ; Server B [general] ; this boxremote IP register = ServerB:very_sec...@79.233.yyy.yyy disallow=all allow=alaw allow=ulaw allow=gsm jitterbuffer=no forcejitterbuffer=no autokill=yes ; the other box [ServerA] type=friend trunk=no auth=md5 encryption=yes secret=very_secret context=from-ServerA qualify=yes host=dynamic ; end of Server B If I replace the WAN addresses of the two routers with addresses on the LAN, everything works. Currently, I am not sure whether it could be a NAT related or Asterisk configuration problem. jg -- _ -- 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 Kotzt Ihr schon, oder kocht Ihr noch? English translation (give or take): Happy Christmas! -- _ -- 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 - Is T.38 possible on SPA8800 FXO port ?
T.38 is a relay protocol for transmitting Group 3 facsimiles over a packet network. It has nothing to do with the analog FXS or FXO signaling, but it should not be too difficult to use Asterisk as a gateway. I never needed to do this myself, so I can't tell whether there are any subtle points. jg -- _ -- 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] odbc connection timeout varable
Why are you concerned? ODBC reconnects automatically if necessary. jg -- _ -- 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] odbc connection timeout varable
It takes a small fraction of second to reconnect. You should not experience any missing info. jg -- _ -- 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] odbc connection timeout varable
Unless of course the database server is not running at all for some reason. But that's not exactly an Asterisk problem. jg -- _ -- 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] odbc connection timeout varable
It should not happen. I have a couple of Asterisk servers using the ODBC connection. I never ever had any problem with ODBC or the database. What database are you using? jg -- _ -- 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] odbc connection timeout varable
So, your DB is not on the same machine? WAN or LAN? jg -- _ -- 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 GOIP Outgoing Callerid not working
You could do a simple pcap trace and analyze the sip protocol to see what is actually happening. You could also look at the sample sip.conf and the file you are using to check whether the Remote Party Identification and possibly P-Asserted identity play a role in the communication with your service provider. jg -- _ -- 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 GOIP Outgoing Callerid not working
One of your jpges showed a combo box close to the bottom with something like CID Forward Mode. I think it said Remote Party ID, which may be part of the problem. Whatever you specify here must be supported by the pbx and I usually look at the pcap traces to check that the necessary headers actually exist. jg -- _ -- 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] allo.com gsm card with AsteriskNOW
Don't they have a kernel module that communicates with the card on one and with DAHDI on the other side? The first steps are probably to check with lspci whether the card is detected and then make sure the allo module is loaded. jg -- _ -- 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] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages
When doing an apt-get upgrade the system packages will be update but sometimes Asterisk is broken. Can you describe what is broken? Which packages do I have to exclude when I do not have time to recompile Asterisk/Dahdi each time? libc? Kernel-Packages? The way things are set up is that you have to compile the wanpipe drivers always against the current versions of the kernel and DAHDI. You could change that with a pretty high risk of breaking things, mainly due to frequent changes in the network related parts of the kernel. I have never seen something break, except for kernel updates, but my systems are RedHat based. I doubt that the Debian based systems are much different as far as interface changes are concerned for a certain release. My guess is that you should be fine if you do not execute apt-get dist-upgrade. jg -- _ -- 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] Ubuntu 12.04 LTS / Asterisk / apt-get upgrade / exclude packages
I believe there is a way in Debian, and therefore probably Ubuntu, to have some script execute automatically following a kernel update. So you might even be able to have DAHDI self-rebuild following a kernel update. I went back to do everything by hand, if necessary. Thorsten and other Wanpipe users need to synchronize DAHDI, the kernel, and the Wanpipe drivers. I don't use Asterisk modules that depend too much on the distro or kernel version, but other setups might have more dependencies. Essentially you can do a kernel update only if the Wanpipe drivers are consistent with the DAHDI and kernel versions and DAHDI works only if the kernel doesn't object, which happens once in a while. The wanpipe drivers rarely (=never) work with the most recent kernels, so running a production machine with the latest beta Fedora or Ubuntu, is something for extreme athletes. jg -- _ -- 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 Phone ( Telecom feature )
You asked this question before and there was an already answer on September 28. jg -- _ -- 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 Phone ( Telecom feature )
Then this may be the wrong forum. Intercom is also a bit vague---there are a couple of different options. Have a look at: http://www.voip-info.org/wiki/view/Asterisk+Paging+and+Intercom You might just have to set the auto-answer feature of a phone, but this would be phone specific. jg -- _ -- 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] Intercom Telephone Feature
Of course, it is possible. Depending on what the desired behavior is, it might suffice to enable the auto-answer feature of an end point. You might also want to read about paging and intercom for different scenarios. jg Dear all, My client has Asterisk based telephony system. He needs to add the intercom feature in his telephones. He has 300 concurrent users with two PRI Channels. I want to check if there is a possibility to have the requested scenario by adding this feature to his current telephone system -- _ -- 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] Multicast AMI?
Hi! Maybe I have overlooked something, but I am sort of facing the following problem. I always used the AMI interface to allow (older) client programs on Windows to use their TAPI client code in order to communicate with Asterisk servers. The functionality is basically minimal as only incoming calls need to get detected and there are occasional outgoing ones. For the outgoing calls, a regular AMI connection is fine, but having a lot of machines ( 20) maintaining separate TCP connection, just to listen to some events seems to be too much effort. Does a (read-only) multicast for relevant events make any sense (except for data protection issues)? I am considering writing a small proxy that does exactly that with correspondig changes for my Telephony Service Provider (TSP). Would it make any sense to implement this as an option for the Asterisk program? Basically, Actions and Action responses via TCP and Events via multicast? jg -- _ -- 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] Special functionality for Secretary/Boss
Why can't you continue within the extension and dispatch whether the call failed or terminated? Simply make a second call. jg -- _ -- 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] (no subject)
Did you start the Asterisk server? jg -- _ -- 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] customizing internal calls
I would let the phone handle the different ring tones, if possible. For my phones a SIPAddHeader with something like Alert-Info: http://127.0.0.1/Ringer3 does the trick, but the syntax might be vendor specific. The other problem should be taken care of with call queues. jg -- _ -- 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] The plain old PBX functionality
If you think it is bad then do not use it; else use it; There is no natural law that requires to publish the sources, even if the software is otherwise free. You can always write your own modules and publish the sources. I have difficulties seeing your point. Without creating a large off-topic response, there is a segment of the open source movement that holds that any software that does not come with source code is bad and should not be touched/used in any fashion by any person/company. -- _ -- 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 performace 64bits
If it's a 64-bit CentOS, then you'll have 64-bit binaries by default. Just compare the size of the binaries with both options. Years ago there could have been occasional problems, if you had 32-bit and 64-bit binaries on your machine. jg -- _ -- 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 Error
Please, show your dial plan and name your Asterisk version. You might be call the Dial application with incomplete arguments. jg Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -- _ -- 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] Get last dialed number in a context?
This is a typical phone feature, but it should be easy to implement this at the pbx level using originate and call files. Actually, I have a robust wakeup call module for hotels that could be used for this. If you need a fast solution you could contact me. jg -- _ -- 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] Get last dialed number in a context?
Wouldn't the various call completion methods require support from the telco? It might be technology dependent and even for the same technology, e.g. ISDN, telcos might not support or enable it. jg -- _ -- 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] Get last dialed number in a context?
Have you tried RetryDial()? This way the receiver will be off-hook all the time, which might be inconvenient. jg -- _ -- 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] wct4xxp Excessive Interrupts Resulting in Unusable System or Card
Just to be sure, what's the output of vmstat 10 10? jg -- _ -- 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] wct4xxp Excessive Interrupts Resulting in Unusable System or Card
Yes, I can see this. Another thing to check would be to start from a different OS (eg from a USB stick) and see how the card behaves on the otherwise same hardware. Since your ProLiant G2 server is almost 10 years old, and the TE410P works with 3.3V only (http://www.digium.com/en/products/telephony-cards/digital/quad-span), it might be worth to check this. jg -- _ -- 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] 2 PRI Card - Interrupt Problem
Yep, that's a known problem with older CentOS version. I have some 6.4 customer machines, which I cannot update easily, so I wrote a little patch for irqbalance (only a path problem). I posted some info here: http://lists.digium.com/pipermail/asterisk-dev/2013-December/064182.html Let me know if you want to look at it. jg -- _ -- 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] deactivate SRTP in asterisk 11
Either you do not compile the srtp module into the Asterisk package or you disable RTP encryption on a phone by phone basis. jg -- _ -- 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] Indications.conf: change volume
The call invitation is only signaled in most cases. You need to check the settings of your phones. Hi, I use Asterisk to create the dial tone (indications.conf), which works quite well. However the generated signal is quite loud at the client side (in comparison to the following speech ). Is there an option to modify the volume? --- Dennis Guse -- _ -- 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] originate woes: extension never executes
How about: [greeting] exten= s,1,Answer() same=n,Background(silence/2hello) same=n,Wait(3) provided you know why you want to call Background() instead of Playback(). jg -- _ -- 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] Two pid's shown in asterisk service status
What does ps aux | grep asterisk say? jg -- _ -- 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] Duplicate incoming channel into two outgoing channels
Wouldn't it make more sense to handle this by just monitoring the calls and doing everything else with normal data processing? jg -- _ -- 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] Duplicate incoming channel into two outgoing channels
What do you mean with voice recorders? Voice mail, if nobody answers, or do want to monitor calls? jg -- _ -- 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] Skip ./configure when source directory has not changed
This wasn't a technical question. It's scam to get some fresh email addresses. jg -- _ -- 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] Skip ./configure when source directory has not changed
I, and possibly others, got some unwanted mail from this thread. Somebody is abusing the email addresses... jg -- _ -- 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] Skip ./configure when source directory has not changed
A silly question bouncing in my head for a long time : when I'm installing-configuring a new Asterisk system, I'm using a script that issue the usual ./configure, make and make install commands to install Asterisk from source. When installation fails for any reason, I would re-run my installation script which in turn, among many things, would launch the above ./configure command. Is there a smart way to accelerate things a bit and skip ./configure when source files have not changed since last configure command was previously run ? Regards You do not need to call ./configure when building the package fails. Just call make, maybe make --jobs=4. If your build fails because of a missing library, then you may (need) to call configure again. jg -- _ -- 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 12 - CDR changes
Can you publish a short stub of your dial plan to see what you are doing? There are the NoCDR, ForkCDR, and ResetCDR applications that might help. jg -- _ -- 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] SIPAddHeader back to source
Hi, I am using the XML-browser and Call-Info header features for some SIP phones. SIPAddHeader(Call-Info: ...) seems to work only in the outgoing direction. Does somebody know a way to send a Call-Info header to the originating SIP device by using only the dial plan? Currently, I am using the XML-browsers to update callee info, but I'd like to use the icon purpose to do that. It's probably easier to embed this functionality into a CTI application using an AMI command like Originate (such that internally Dial() gets called twice), but this triggers it from the outside. sipsak could be called from extensions.conf, but I'd like to avoid that. Transferring to a Local channel after entering the dial plan might also work, but that looks clumsy. I am sorry if I have overlooked a standard method to send a header back to the source. jg -- _ -- 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] dahdi + dlink du128ta
I do have a usb ISDN modem that I would like to make it work with dahdi. is it possible? I am running debian 7, with dahdi 2.9, asterisk 11.8 dahdi cannot find it at the moment, unless there is something else to be done. I do not think so. If your very old ISDN modem uses a Cologne Chip (HFC), then there might be a very small chance to make it work with mISDN. jg -- _ -- 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] res-odbc sanity check reconnecting
I would appreciate if someone could help me with the following issue: http://pastebin.com/bTskMLVw My res_odbc.conf file look as follows: http://pastebin.com/bhReQkXQ Nothing to really worry about. The ODBC driver automatically reconnects to MySQL as the system already told you. jg -- _ -- 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] res-odbc sanity check reconnecting
I'm getting a lot of errors in CLI and this become annoying. And when I use database from localhost, I don't have such an issue. So I don't understand how to identify the reason of this problem. First, please do not add your signature in the current form to your posts. It is irrelevant here and it takes too much time to delete them... You did not mention that the db is running on a different host. You could publish your odbc.ini, but essentially you need to find out whether there is a problem with your LAN or the way you have configured things. Start a pcap trace and look for delays related to dns, etc... It is likely that your problem is not related to Asterisk. jg -- _ -- 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] CONNECTEDLINE(name) ISDN problem
When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I see that CONNECTEDLINE(num) maps to the connected number information element and CONNECTEDLINE(name) to the display element. The pcap trace does actually contain my CONNECTEDLINE(name) plus a leading byte with a value of 0xB1 (or \261 in octal notation). This additional byte is part of the announced string length. Now I wonder, whether this byte is causing the trouble. Does anybody know what this leading byte is actually doing there? jg -- _ -- 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
