Re: [asterisk-users] Mailing a fax with mutt does not succeed
Hi
Login as user asterisk and then try it
e.g. sudo su asterisk
and then try to execute the command.
Last time I had this issue I had to set the shell environment for
asterisk with the
chsh
command
On Wed, 2013-06-19 at 13:03 -0500, Daniel - Asterisk wrote:
Hello everyone,
I'm trying to send a received fax with mutt, when I try it from the
Linux shel it works, but when trying with Asterisk's System command it
doesn't.
Successful Linux command:
echo | mutt -s New fax earohua...@gmail.com
-a /tmp/faxes/20130619.tif
Unsuccessful Asterisk Command:
same = n,System(mutt -s New fax elder.arohua...@gmail.com -a
${FAXDEST}/${tempfax}.tif)
I'm using Asterisk 1.8.19.0 on Debian 6.0.6, Asterisk was installed by
root.
Any hint will be appreciated.
Elder D. Arohuanca
Lima - Peru
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--
Ishfaq Malik i...@pack-net.co.uk
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552
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[asterisk-users] Customer src in CDR with incoming sipp calls
Hello,
I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1.
I've dedicated a context to sipp in my dialplan.
Everything works OK expect that calls from sipp comes in with a CallerID
set to sipp and this sipp value is stored in CDR.
1. I can change the value of the CallerID but how can I have the calls from
sipp traced in CDR with a customized src field value ?
[from-sipp]
exten = _X., 1, Whatever()
same = n, set(CallerID(num)=987654321))
same = n, Goto(maincontext, 123456789, 1)
[maincontext]
exten = _X., 1, Whatever()
same = n, Dial(SIP/foobar/${EXTEN})
I've read in CDR function online doc that src field is Readonly so I can't
simple use Set(CDR(src)=987654321).
Would a combination of NoCDR, ForkCDR or whatever help ?
2. Beside that, is there a what to set sipp CallerID from command line ?
Doc mentions -cdr_str option but I couldn't get anything with it.
Regards
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Re: [asterisk-users] no silk translation ?
On 11.06.2013, at 0:24, Sean Darcy wrote: Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: Silk is enabled only after asterisk restart. for silk work need codecs.conf with silk configuration res_format_attr_silk.so - loaded codec_silk.so - loaded please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Codecs+and+Audio+Formats [Jun 10 16:18:22] WARNING[4090][C-000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng- to Motif/+12025551...@voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make SIP/ng- compatible with Motif/+12025551...@voice.google.com-da3c == Spawn extension (BaseDial, s, 4) exited non-zero on 'SIP/ng-' core show translations doesn't include any SILK. SILK is installed: core show codec 100018 100018 SILK Custom Format 8khz 100018 SILK Custom Format 12khz 100018 SILK Custom Format 16khz 100018 SILK Custom Format 24khz sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip video endpoint with asterisk
hi, i need some small sip video endpoint for cloud videoconference (like bluejeans) i have this idea VIDEO OUT TV or projector with HDMI VIDEO IN cameras with h264 hw enconding - http://downloads.element14.com/raspberry-pi-camera/ http://downloads.element14.com/raspberry-pi-camera/ - logitech C920 - Creative Live! Cam Connect HD - ??? ENDPOINT - raspberry - miniPC linux + asterisk ? https://wiki.asterisk.org/wiki/display/AST/Video+Console https://wiki.asterisk.org/wiki/display/AST/Video+Console AUDIO IN + AUDIO OUT microphone with integrated speakers for the table http://www.jabra.com/Products/PC_Headsets/Jabra_SPEAK__510_Series/Jabra_Speak_510 http://www.jabra.com/Products/PC_Headsets/Jabra_SPEAK__510_Series/Jabra_Speak_510 (bluetooth connection!!!) http://www.phnxaudio.com/quattro3 http://www.phnxaudio.com/quattro3 http://www.yamaha.com/products/en/communication/usb_conference_speakerphones/ http://www.yamaha.com/products/en/communication/usb_conference_speakerphones/ http://www.dev-audio.com/products/microcone/ http://www.dev-audio.com/products/microcone/ http://www.clearone.com/products_chat160 http://www.clearone.com/products_chat160 do you think it is possible? any recommendations? -- --- Marek Cervenka === -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] packet counts: twice as high on one leg?
Hi all, I have two phones that I've been comparing (different manufacturers). To debug call quality issues on one of them, I've been using calls from the phone to our main DID, so 3 SIP sessions exist (phone asterisk then asterisk provider, and the providerasterisk for the DID). The bad phone shows roughly twice the number of packets on the phoneasterisk session as on the other two sessions. The good phone shows roughly equal packet counts on each of the 3 sessions. I've used asterisk server MTUs of 1440 and 1500, but it makes no difference. Is this double-packet-count a clue, a problem, or a red herring? The packet counts are shown using sip show channelstats. - Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] packet counts: twice as high on one leg?
Have you checked whether the same codecs, or codecs with the same bandwidth requirements, are used? 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] packet counts: twice as high on one leg?
On 06/20/2013 11:56 AM, jg wrote: Have you checked whether the same codecs, or codecs with the same bandwidth requirements, are used? Yes, it is ulaw everywhere. - Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] packet counts: twice as high on one leg?
On 06/20/2013 11:56 AM, jg wrote: Have you checked whether the same codecs, or codecs with the same bandwidth requirements, are used? Here's an example of a simple outgoing call. Everything is ulaw. The 192.x.x.x phone has roughly twice the packet count of the provider session. The lost packet count is nonsensical on one session. Sigh. - Mike steerpike*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter 209.217.98.130 0c15efc03f2 00:01:03 003069 104829 (97.16%) 0. 003040 00 ( 0.00%) 0.0002 192.168.0.36 qY0p292XeDl 00:01:03 006121 00 ( 0.00%) 0. 006096 00 ( 0.00%) 0.0001 2 active SIP channels steerpike*CLI sip show channels Peer User/ANR Call ID Format Hold Last MessageExpiry Peer 209.217.98.130 6139419467 0c15efc03f243c7 (ulaw) No Tx: ACK6136866675 192.168.0.36 mjc_office qY0p292XeDlPcLk (ulaw) No Rx: ACKmjc_office steerpike*CLI core show version Asterisk 11.4.0 built by root @ steerpike.avtechpulse.com on a x86_64 running Linux on 2013-06-19 12:10:47 UTC -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] packet counts: twice as high on one leg?
On Thu, 20 Jun 2013, Dr. Michael J. Chudobiak wrote: Here's an example of a simple outgoing call. Everything is ulaw. The 192.x.x.x phone has roughly twice the packet count of the provider session. Would running wireshark yield any clues? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] packet counts: twice as high on one leg?
There's one thing on my list of things to check, that could be relevant for your problem, too. Packet count is one thing, transferred data is another one. If one phone uses smaller UDP packages, then the packet count should increase in reciprocally. I have read some comments on the net that smaller packages are preferable because lost packages have less impact on the hearable audio. Getting a complete pcap trace with subsequent wireshark analysis is probably a tedious but appropriate way to analyse this. Currently I don't know what determines the size of the UDP packages and whether there are any general ways to control this size. 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] packet counts: twice as high on one leg?
Packet count is one thing, transferred data is another one. If one phone uses smaller UDP packages, then the packet count should increase in reciprocally. I have read some comments on the net that smaller packages are preferable because lost packages have less impact on the hearable audio. Aha. I overlooked that some phones had ulaw:10 in sip.conf, instead of the standard ulaw:20. That explains the packet count difference. It seems my call quality issues are coming from something else. - Mike -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 / PHP-AGI / pthreads
On Thu, 20 Jun 2013, Satish Barot wrote:
Would you mind sharing a sample of your pthread-ed C AGI? This will help
someone like me who has written AGI in Perl/PHP and now exploring C AGI.
The source code for this particular AGI is about 600 lines and uses my own
AGI library (written before other C AGI libraries were available) so I
don't think it has a whole lot of value to other programmers.
First a little background on why I used pthreads in this AGI...
The application was an adult chat platform. (Mostly) guys call in to chat
to (mostly) women and pay $1 to $4 per minute. At the end of the call, the
accumulated charges are billed to their their credit card.
Charges to credit cards usually take 2 transactions. The first transaction
('OPEN') places a temporary 'hold' on the card for $XXX. At the end of the
call, a second transaction ('SALE') finalizes the billing for the actual
charges for the call.
(This '2 step' process is why you may have trouble charging an expensive
dinner after checking into a hotel -- the hotel may have placed a 'hold'
against your credit limit for the expected charge for your entire stay.)
As originally implemented, Asterisk would play a prompt asking the caller
to enter their card details. After the caller entered a 'valid' card
number (that passes Luhn mod 10) and expiration date (between the current
month current year and current year + 7). My AGI would then play a
prompt ('Please wait while your card is being verified') and then transmit
the card details to our card processor. Getting a response from the card
processor takes a second or so. My AGI returned the card processor
response code as the 'priority' so the dialplan could continue to the main
menu or play a prompt indicating why the card details failed. After 3
failures, a caller was played a prompt instructing them on alternative
billing methods (high rate NPAs, check by phone, etc).
This client was extremely particular about the 'caller experience' and
thought the 'second or so' of silence while the card was being verified
was excessive and had to be eliminated.
Since the card processing time was not under my control, my solution was
to change my AGI to create a second thread to play the 'please wait'
prompt while the 'main line' code shipped off the card details to the
processor.
When I got the processor's response (which was usually before the prompt
finished playing), the 'main line' code would 'join' the prompt thread. If
the prompt thread was finished playing the prompt, execution continued
immediately. If the prompt thread was still playing the prompt, execution
continued as soon as the prompt thread received the AGI response from
Asterisk. None of this took any special programming or synchronizing from
me. It all 'just worked' because of the way pthreads are implemented.
So, what did I learn that would be of value to you? Really understand the
AGI protocol and you won't make a bunch of silly mistakes.
The AGI protocol is very simple:
1) Read the AGI environment from stdin before you do anything in your AGI
that interacts with Asterisk.
2) Write your AGI request to stdout.
3) Read Asterisk's AGI response from stdin.
4) Rinse, lather, repeat (steps 2 3).
If you use an established AGI library (like you really, really should),
this should be taken care of automagically, but understanding what's
really going on will help when things don't work as expected.
The number 1 mistake new AGI programmers make is not reading the AGI
environment. Sometimes they 'get away' with it, some times they don't.
The number 2 mistake is not reading the responses (step 3). Again,
sometimes it works, sometimes it doesn't
The number 3 mistake new (and old salts like me still make on occasion) is
doing any I/O on stdin or stdout. A lot of bad debugging centers around
'throw in a printf somewhere to see what's going on' instead of using
'real programmer' tools like gdb*.
If you're using an established AGI library, the crucial relationship
between steps 2 and 3 will be take care of -- unless you're using multiple
threads.
If thread 1 issues an AGI request (writing to stdout) and then thread 2
issues an AGI request (also writing to stdout) before thread 1 reads it's
response, you've broken the protocol and bad things will happen. Maybe not
immediately, but certainly when you're demonstrating the system to your
boss.
So my first suggestion (after using an established library and without
knowing what your use case is) would be to 'designate' a single thread as
'the Asterisk' thread and only interact (via stdin and stdout) with
Asterisk in that thread. If that's not feasible, you're going to need some
sort of 'semaphore' mechanism so you don't 'intermingle' your AGI requests
and responses and thereby violate the AGI protocol.
Another couple of suggestions unrelated to threads, but are best (IMNSHO)
practices for AGIs in general:
1) Use getopt_long().
Re: [asterisk-users] packet counts: twice as high on one leg?
Aha. I overlooked that some phones had ulaw:10 in sip.conf, instead of the standard ulaw:20. That explains the packet count difference. It seems my call quality issues are coming from something else. ... and this explains how to set the packet size. Answer to get answers, or so. 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] Questions about sRTP
Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said: == *Note: There is no optional SRTP mode in Asterisk, i.e. if encryption is active on peer, it will not accept non-ciphered audio and viceversa. On the IP phones, however, it is possible to have unsecure calls if the other peer does not support SRTP, i.e. incoming calls may work, but not outgoing calls. This is an Asterisk limitation (Snom supports also the “optional”mode on SRTP sending two m=audio attributes, but Asterisk does not know how to handle those descriptors).* == This is from a quite dated article (2011), so I'm hoping that I newer versions of Asterisk will fall back on plaintext if TLS isn't available for some reason. Secondly, is there any way to detect if a call is secure from inside the dialplan or AGI script? I think that's all for now. Thanks in advance, Mike Diehl. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Exceptionally long queue length (help!)
Help! I have providers configured that send me incoming calls via SIP. There's one that seems to make trouble. As soon as I get a few concurrent calls through this peer, Asterisk CPU load goes up to 250%, audio becomes laggy and I get hundreds of these per second in the logs: [Jun 20 20:00:16] WARNING[1206][C-0001] channel.c: Exceptionally long queue length queuing to SIP/provider11-0001 And the following is what I always get every few seconds for every call via this peer. I think this only happens when I'm answering the call in Asterisk itself, i.e. right now I'm just streaming MusicOnHold to callers. [Jun 20 20:00:22] NOTICE[1206][C-0001] channel.c: Dropping incompatible voice frame on SIP/provider11-0001 of format ulaw since our native format has changed to (alaw) Peer config [provider11] host=121.16.15.230 type=peer insecure=port,invite context=provider-in disallow=all allow=alaw canreinvite=no dtmfmode=rfc2833 Before I had allow=ulaw allow=alaw and removed the ulaw codec hoping that the above NOTICEs might disappear, but without success. Asterisk 11.4.0-1 from the official repository under CentOS 6.4 64bit. Help! :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about media before connect
I need to block any audio before there is a connect, in SIP. How do I tell the DIAL application to behave like that? Yours Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 need a second opinion on a new phone system deployment
Please also have a look at the gateway boxes from berofix (http://wiki.beronet.com/index.php/Main_Page). I am not affiliated but have used different products from them over last few yeas and all have survived and are stable. Documentation is open and free on their wiki. They provide updates. They are not the cheapest but they have different vendors and they are sold in online webshops. You can choose for the inside PCI(e) cards or their external boxes. Last few years I went for the external boxes. They can be fitted in a server rack or you mount them against the wall with screws. Regards, Michel. On 16-06-13 16:55, Nunya Biznatch wrote: Thanks again to everyone that's responded thus far. I have once again bundled the questions and answers into a single email, and am responding below. On 6/14/2013 9:43 AM, Nunya Biznatch wrote: Howdy All, They say opinions are like belly buttons, everybody has one. (that's the clean version of the saying). So I'm asking for yours. I hope you see it as a fun exercise. I'm designing a phone system from the ground up. Will be about 1000-1300 seats mixed 80/20 VoIP/Analog. 58-acre campus environment with 23 buildings. Userbase is emergency services organization, 24/7/365 operation. Down time is not an option, but blips are acceptable. Repair time is immediate. We need failover for the failover essentially. However, money is a major factor, so I have to do it all for nothing. So here's what I'm thinking. Please throw in your 2 cents. Network will be separate for phones. Fiber infrastructure available between buildings as well as copper. Internet access will be limited to a single administrative console on a temporary basis, and then only when remote 3rd party support is required. Access for 3rd party support will be supervised through remote access tools such as VNC, GoToMeeting, etc... etc... System will have zero access to local data network. This means all ancillary support servers such as DHCP, DNS, NTP, FTP, etc...etc... will be specific to the phone system. Yes, I know some responders at this time will become fixated on me gaining this connectivity. It ain't gonna happen. It's not an option. Period, end of story. These are the parameters I must work within. Trying to fix that will be a non-starter. The phone system will upgrade an existing TDM-based system. Mitel SX2000 with NuPoint Voicemail. This will not be a dump-trunk replacement. I expect at least a one to two-year transition, meaning we will have time to find problems, work bugs, and learn over time, with minimized impacts. It also means we'll be supporting two systems for some time. PBX is 97% serving your basic phone on the desk. Nothing special. Customers expect the usual list of features. There will be a goodly number of hints required for BLF on maybe 150 phones. There is one office of about 30 phones in a call-center environment that will need that service. They would be considered low volume (but don't tell them that). My Skills... I am not a Linux kung fu master, but I have built and managed my share of Linux servers on mutiple Linux flavors. I am a DCAA, having been through formal training, and have been playing with Asterisk for years, but always in fits and spurts and never in a live environment so I am by no means a kung fu master there either. I have started dabbling with virtualizations via XEN, but I am not comfortable enough with it to go live this first round. I can see myself implementing it in about three years once we're totally comfortable with what we have, so I can then have time to get that skill sorted. I was a network engineer for the US no3. telecom for a number of years, 10-years in comm-electronics in the military before that. Telecom my entire career. I've got the kung-fu to handle the network side of the house, and having administrated multiple PBXs for decade-plus, I've got the concepts down. No plans to build databases for things like directories, etc... I'm not greatly confident in those skills, and to date, haven't found anything that really stands out that would make me require that. You may think otherwise, so please chime in. I say that, but at the same time I recognize I may require a GUI interface once fully deployed to allow lower-skilled people to follow the motions to complete simple moves, adds, and changes. I'm fighting the uphill battle that is the GUI is new, CLI is old mentality. System will use G.722 for VoIP Phones. So there's the groundwork. Here's the hardware plan. Plan is to build my own servers following industry standards (ATX) and using industry standard equipment. Why? Spares? Whether redundant or not, I will still have spares for the most common elements on the shelf so equipment can be returned to service as quickly as possible. This will also allow me to be comfortable with more basic server configurations and help keep cost down. For example, Servers with single
Re: [asterisk-users] Questions about sRTP
Mike Diehl wrote: Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said: Your statement is incorrect. Asterisk supports TLS as an optional signaling transport (although if you do SDES SRTP without it then someone can snoop on your keys and ultimately decrypt your media). What it does not support is optional *SRTP*. If a device requests SRTP and it's not possible, the call will fail. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mailing a fax with mutt does not succeed
Hello jg:
When mutt is called from Asterisk's dialplan there's no output at mail.log
When I use:
echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax} /tmp/ocurrencias.txt 21
replacing FAXDEST and TEMPFAX with proper values, the output is as follows:
Jun 20 16:16:16 SERVER-NAME sendmail[21276]: My unqualified host name
(SERVER-NAME) unknown; sleeping for retry
Jun 20 16:17:16 SERVER-NAME sendmail[21276]: unable to qualify my own
domain name (SERVER-NAME) -- using short name
Jun 20 16:17:16 SERVER-NAME sendmail[21276]: r5KLHGgk021276: from=root,
size=116501, class=0, nrcpts=1, msgid=20130620211615.GA21267@SERVER-NAME,
relay=root@localhost
Jun 20 16:17:17 SERVER-NAME sm-mta[21285]: r5KLHGNY021285:
from=root@SERVER-NAME, size=116646, class=0, nrcpts=1,
msgid=20130620211615.GA21267@SERVER-NAME, proto=ESMTP, daemon=MTA-v4,
relay=localhost [127.0.0.1]
Jun 20 16:17:17 SERVER-NAME sendmail[21276]: r5KLHGgk021276: to=
earohua...@gmail.com, ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:01,
mailer=relay, pri=146501, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0,
stat=Sent (r5KLHGNY021285 Message accepted for delivery)
Jun 20 16:17:19 SERVER-NAME sm-mta[21287]: STARTTLS=client, relay=
gmail-smtp-in.l.google.com., version=TLSv1/SSLv3, verify=FAIL,
cipher=RC4-SHA, bits=128/128
Jun 20 16:17:20 SERVER-NAME sm-mta[21287]: r5KLHGNY021285: to=
earohua...@gmail.com, ctladdr=root@SERVER-NAME (0/0), delay=00:00:03,
xdelay=00:00:03, mailer=esmtp, pri=236646, relay=gmail-smtp-in.l.google.com.
[173.194.76.26], dsn=2.0.0, stat=Sent (OK 1371763040 f6si834075qaf.111 -
gsmtp)
ocurrencias.txt is empty also.
Elder Arohuanca
On Wed, Jun 19, 2013 at 3:12 PM, jg webaccou...@jgoettgens.de wrote:
More things to try:
(1) Is there any entry in /var/log/maillog (or equivalent log file)? If
so, mutt basically works and the messages should give some clues.
(2) What happens if you call mutt without any attachments?
I am using mutt in exactly the same way and it works.
jg
Am 19.06.2013 21:50, schrieb Daniel - Asterisk:
Hi Andre:
I added echo to provide STDIN, I'm sure on variable contents, please see
bellow
Hello Steve,
1. I've just addd echo at my sentence, please see output bellow.
2. Asterisk is executing as root, I think Asterisk has access to read TIF
files since I've used ls, chmod, cp mv from Asterisk's CLI with '!'
character.
3. I don't get you, please give some advice to try using Verbose instead
System
4. I don't know how to get this, but I'm using /usr/bin/mutt as you can
see bellow.
5. I have redirected output of System this way : System(echo |
/usr/bin/mutt -s New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax}
/tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
DIALPLAN:
[ Context 'default' created by 'pbx_config' ]
'*95' = 1. NoOp(trying to send a fax to an email)
2. Set(FAXDEST=/tmp/faxes)
3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11})
4. NoOp(file name is: ${tempfax})
5. Goto(incoming-fax,fax,7)
[ Context 'incoming-fax' created by 'pbx_config' ]
'fax' = 1. Verbose(3,Incoming fax)
...
5. ReceiveFax(${FAXDEST}/${tempfax})
6. Verbose(3,- Fax receipt completed with status:
${FAXSTATUS})
7. System(echo | /usr/bin/mutt -s New fax
earohua...@gmail.com -a ${FAXDEST}/${tempfax})
8. NoOp(System command status is: ${SYSTEMSTATUS})
9. Hangup()
ASTERISK CLI OUTPUT:
-- Goto (default,*95,1)
-- Executing [*95@default:1] NoOp(SIP/40106-1ea1, trying to
send a fax to an email) in new stack
-- Executing [*95@default:2] Set(SIP/40106-1ea1,
FAXDEST=/tmp/faxes) in new stack
-- Executing [*95@default:3] Set(SIP/40106-1ea1,
tempfax=20130619.tif) in new stack
-- Executing [*95@default:4] NoOp(SIP/40106-1ea1, file name
is: 20130619.tif) in new stack
-- Executing [*95@default:5] Goto(SIP/40106-1ea1,
incoming-fax,fax,7) in new stack
-- Goto (incoming-fax,fax,7)
-- Executing [fax@incoming-fax:7] System(SIP/40106-1ea1, echo
| /usr/bin/mutt -s New fax earohua...@gmail.com -a
/tmp/faxes/20130619.tif) in new stack
-- Executing [fax@incoming-fax:8] NoOp(SIP/40106-1ea1, System
command status is: APPERROR) in new stack
-- Executing [fax@incoming-fax:9] Hangup(SIP/40106-1ea1, ) in
new stack
Elder D. Arohuanca
Lima - Peru
On Wed, Jun 19, 2013 at 1:38 PM, Andre Courchesne voipfor...@gmail.comwrote:
Why echo | ?
Alsy are you sire of the content of ${FAXDEST} and ${tempfax}.
Add some NoOp before.
On 2013-06-19, at 2:29 PM, Daniel - Asterisk earohua...@gmail.com
wrote:
Hi Andre,
I've tried with:
System(echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax})
with no success, value of SYSTEMSTATUS variable is
Re: [asterisk-users] Mailing a fax with mutt does not succeed
Have you tried calling a bash script that in turns calls mutt. That way you
could debug much easier, adding echo to a log file.
Sent from my iPhone
On 2013-06-20, at 5:27 PM, Daniel - Asterisk earohua...@gmail.com wrote:
Hello jg:
When mutt is called from Asterisk's dialplan there's no output at mail.log
When I use:
echo | /usr/bin/mutt -s New fax earohua...@gmail.com -a
${FAXDEST}/${tempfax} /tmp/ocurrencias.txt 21
replacing FAXDEST and TEMPFAX with proper values, the output is as follows:
Jun 20 16:16:16 SERVER-NAME sendmail[21276]: My unqualified host name
(SERVER-NAME) unknown; sleeping for retry
Jun 20 16:17:16 SERVER-NAME sendmail[21276]: unable to qualify my own domain
name (SERVER-NAME) -- using short name
Jun 20 16:17:16 SERVER-NAME sendmail[21276]: r5KLHGgk021276: from=root,
size=116501, class=0, nrcpts=1, msgid=20130620211615.GA21267@SERVER-NAME,
relay=root@localhost
Jun 20 16:17:17 SERVER-NAME sm-mta[21285]: r5KLHGNY021285:
from=root@SERVER-NAME, size=116646, class=0, nrcpts=1,
msgid=20130620211615.GA21267@SERVER-NAME, proto=ESMTP, daemon=MTA-v4,
relay=localhost [127.0.0.1]
Jun 20 16:17:17 SERVER-NAME sendmail[21276]: r5KLHGgk021276:
to=earohua...@gmail.com, ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:01,
mailer=relay, pri=146501, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent
(r5KLHGNY021285 Message accepted for delivery)
Jun 20 16:17:19 SERVER-NAME sm-mta[21287]: STARTTLS=client,
relay=gmail-smtp-in.l.google.com., version=TLSv1/SSLv3, verify=FAIL,
cipher=RC4-SHA, bits=128/128
Jun 20 16:17:20 SERVER-NAME sm-mta[21287]: r5KLHGNY021285:
to=earohua...@gmail.com, ctladdr=root@SERVER-NAME (0/0), delay=00:00:03,
xdelay=00:00:03, mailer=esmtp, pri=236646, relay=gmail-smtp-in.l.google.com.
[173.194.76.26], dsn=2.0.0, stat=Sent (OK 1371763040 f6si834075qaf.111 -
gsmtp)
ocurrencias.txt is empty also.
Elder Arohuanca
On Wed, Jun 19, 2013 at 3:12 PM, jg webaccou...@jgoettgens.de wrote:
More things to try:
(1) Is there any entry in /var/log/maillog (or equivalent log file)? If so,
mutt basically works and the messages should give some clues.
(2) What happens if you call mutt without any attachments?
I am using mutt in exactly the same way and it works.
jg
Am 19.06.2013 21:50, schrieb Daniel - Asterisk:
Hi Andre:
I added echo to provide STDIN, I'm sure on variable contents, please see
bellow
Hello Steve,
1. I've just addd echo at my sentence, please see output bellow.
2. Asterisk is executing as root, I think Asterisk has access to read TIF
files since I've used ls, chmod, cp mv from Asterisk's CLI with '!'
character.
3. I don't get you, please give some advice to try using Verbose instead
System
4. I don't know how to get this, but I'm using /usr/bin/mutt as you can see
bellow.
5. I have redirected output of System this way : System(echo |
/usr/bin/mutt -s New fax earohua...@gmail.com -a ${FAXDEST}/${tempfax}
/tmp/ocurrencies.txt 21), ocurrencies.txt is empty.
DIALPLAN:
[ Context 'default' created by 'pbx_config' ]
'*95' = 1. NoOp(trying to send a fax to an email)
2. Set(FAXDEST=/tmp/faxes)
3. Set(tempfax=${SHELL(ls /tmp/faxes/*.tif):11})
4. NoOp(file name is: ${tempfax})
5. Goto(incoming-fax,fax,7)
[ Context 'incoming-fax' created by 'pbx_config' ]
'fax' = 1. Verbose(3,Incoming fax)
...
5. ReceiveFax(${FAXDEST}/${tempfax})
6. Verbose(3,- Fax receipt completed with status:
${FAXSTATUS})
7. System(echo | /usr/bin/mutt -s New fax
earohua...@gmail.com -a ${FAXDEST}/${tempfax})
8. NoOp(System command status is: ${SYSTEMSTATUS})
9. Hangup()
ASTERISK CLI OUTPUT:
-- Goto (default,*95,1)
-- Executing [*95@default:1] NoOp(SIP/40106-1ea1, trying to send
a fax to an email) in new stack
-- Executing [*95@default:2] Set(SIP/40106-1ea1,
FAXDEST=/tmp/faxes) in new stack
-- Executing [*95@default:3] Set(SIP/40106-1ea1,
tempfax=20130619.tif) in new stack
-- Executing [*95@default:4] NoOp(SIP/40106-1ea1, file name is:
20130619.tif) in new stack
-- Executing [*95@default:5] Goto(SIP/40106-1ea1,
incoming-fax,fax,7) in new stack
-- Goto (incoming-fax,fax,7)
-- Executing [fax@incoming-fax:7] System(SIP/40106-1ea1, echo |
/usr/bin/mutt -s New fax earohua...@gmail.com -a
/tmp/faxes/20130619.tif) in new stack
-- Executing [fax@incoming-fax:8] NoOp(SIP/40106-1ea1, System
command status is: APPERROR) in new stack
-- Executing [fax@incoming-fax:9] Hangup(SIP/40106-1ea1, ) in
new stack
Elder D. Arohuanca
Lima - Peru
On Wed, Jun 19,
Re: [asterisk-users] Mailing a fax with mutt does not succeed
Hi Elder! I am currently busy with a problem of one of my customers. I am pretty sure that your setup requires only minor changes to make things work. I have several setups that use mutt exactly in the same way as you are trying to do (except that I am using postfix instead of sendmail, but this shouldn't matter). I'll have a look at your log entries tomorrow. 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] Questions about sRTP
On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote: Mike Diehl wrote: Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said: Your statement is incorrect. Asterisk supports TLS as an optional signaling transport (although if you do SDES SRTP without it then someone can snoop on your keys and ultimately decrypt your media). What it does not support is optional *SRTP*. If a device requests SRTP and it's not possible, the call will fail. So then, is it safe to say that Asterisk will ALLOW a secure phone call, but the client hast to REQUEST it? I understand that requesting SRTP without SIP/TLS is evil; I just misunderstood what I was reading. I'm also thinking that the AGI script I use to route calls can check if either leg of a call comes from or goes to port 5061 and play a sound file to indicate that the cal is 'secure.' Does this seem reasonable? Thanks, Mike. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk -rx core show channels + time
When I type: asterisk -rx core show channels I usually get Channel Location State Application(Data) SIP/pstn--03 7807574622@internal: Up Dial(SIP/77807574622@pstn-9998 SIP/pstn-9998-03 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 -rx core show channels + time
Hi You can do, core show channels verbose Kind Regards On Thu, Jun 20, 2013 at 6:45 PM, Joseph syscon...@gmail.com wrote: When I type: asterisk -rx core show channels I usually get Channel Location State Application(Data) SIP/pstn--03 7807574622@internal: Up Dial(SIP/77807574622@pstn-9998 SIP/pstn-9998-03 (None) Up AppDial((Outgoing Line)) Is there a way to pull information about time the channel started? -- Joseph -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Simple support on Asterisk 11
Eloi, My responses are inline. Thanks a lot for this detailed answer : You're welcome. Thank you for responding. A lot of people forget to do so and future list readers are left wondering whether or not the proposed solution worked. - I managed to have it working disabling auth message request : auth_message_requests = no in sip.conf - pedantic=no does not resolve the issue - reenabling auth_message_requests = yes and removing pedantic option, your patch in chan_sip resolves the issues ! As it looks like pidgin has an issue, I guess that we can use it as a workaround. I'm glad my patch worked but keep in mind that it really is a workaround, because it will cause actual retransmits of MESSAGE requests to be treated as new requests. This may not cause you any issues, but the root problem should still be addressed by submitting a report to the Pidgin bug tracker [1]. It's a straightforward problem so if you provide a link to your original post [2] the developers should be able to resolve it quickly. I would like know to enable presence notification between each users. To fulfill it, I am using http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html Am I doing it in a good way ? Yes, but there is a 4th edition of Asterisk: The Definitive Guide available in the Open Feedback Publishing System [3] that is focused on documenting Asterisk 11. [1] https://developer.pidgin.im/wiki/TipsForBugReports [2] http://lists.digium.com/pipermail/asterisk-users/2013-June/279569.html [3] http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queue Frame
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?
Regards,
Gopal.
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Re: [asterisk-users] Asterisk Queue Frame
On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?
It looks like you are getting the Exceptionally long queue length warning
message. The
change you mention will just increase the allowed size of the queue. Doing
that won't really
help much as it will just delay getting the message. That warning message
means Asterisk
is falling behind in processing frames on the channel.
Richard
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Re: [asterisk-users] Asterisk Queue Frame
actually when i get the message my call volume is around 180 to 200
calls will that matter... and some calls being transferred to other
location and some are making outbound calls, some are inbound...
Is there any way for optimization?
On Fri, Jun 21, 2013 at 5:57 AM, Richard Mudgett rmudg...@digium.comwrote:
On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?
It looks like you are getting the Exceptionally long queue length
warning message. The
change you mention will just increase the allowed size of the queue.
Doing that won't really
help much as it will just delay getting the message. That warning message
means Asterisk
is falling behind in processing frames on the channel.
Richard
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Re: [asterisk-users] Questions about sRTP
On Thu, Jun 20, 2013 at 5:10 PM, Mike Diehl mdiehlena...@gmail.com wrote: On Thu, Jun 20, 2013 at 2:05 PM, Joshua Colp jc...@digium.com wrote: Mike Diehl wrote: Hi all, I'm getting ready to setup SIP/TLS and SRTP. But I have a few questions. The first one is that I was reading an article at: https://supportforums.cisco.com/docs/DOC-15381 That indicated that Asterisk doesn't support TLS as an OPTIONAL transport. It's either all or nothing. Specifically, this is what it said: Your statement is incorrect. Asterisk supports TLS as an optional signaling transport (although if you do SDES SRTP without it then someone can snoop on your keys and ultimately decrypt your media). What it does not support is optional *SRTP*. If a device requests SRTP and it's not possible, the call will fail. So then, is it safe to say that Asterisk will ALLOW a secure phone call, but the client hast to REQUEST it? I understand that requesting SRTP without SIP/TLS is evil; I just misunderstood what I was reading. I'm also thinking that the AGI script I use to route calls can check if either leg of a call comes from or goes to port 5061 and play a sound file to indicate that the cal is 'secure.' Does this seem reasonable? You can query a channel using the CHANNEL function ( https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL) to see if the channel currently supports secure communication, and you can request that the outbound channel be made secure using the same function. An example of doing this is on the wiki: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / PHP-AGI / pthreads
On Thu, Jun 20, 2013 at 10:54 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Thu, 20 Jun 2013, Satish Barot wrote:
Would you mind sharing a sample of your pthread-ed C AGI? This will help
someone like me who has written AGI in Perl/PHP and now exploring C AGI.
The source code for this particular AGI is about 600 lines and uses my own
AGI library (written before other C AGI libraries were available) so I
don't think it has a whole lot of value to other programmers.
First a little background on why I used pthreads in this AGI...
The application was an adult chat platform. (Mostly) guys call in to chat
to (mostly) women and pay $1 to $4 per minute. At the end of the call, the
accumulated charges are billed to their their credit card.
Charges to credit cards usually take 2 transactions. The first transaction
('OPEN') places a temporary 'hold' on the card for $XXX. At the end of the
call, a second transaction ('SALE') finalizes the billing for the actual
charges for the call.
(This '2 step' process is why you may have trouble charging an expensive
dinner after checking into a hotel -- the hotel may have placed a 'hold'
against your credit limit for the expected charge for your entire stay.)
As originally implemented, Asterisk would play a prompt asking the caller
to enter their card details. After the caller entered a 'valid' card number
(that passes Luhn mod 10) and expiration date (between the current month
current year and current year + 7). My AGI would then play a prompt
('Please wait while your card is being verified') and then transmit the
card details to our card processor. Getting a response from the card
processor takes a second or so. My AGI returned the card processor response
code as the 'priority' so the dialplan could continue to the main menu or
play a prompt indicating why the card details failed. After 3 failures, a
caller was played a prompt instructing them on alternative billing methods
(high rate NPAs, check by phone, etc).
This client was extremely particular about the 'caller experience' and
thought the 'second or so' of silence while the card was being verified was
excessive and had to be eliminated.
Since the card processing time was not under my control, my solution was
to change my AGI to create a second thread to play the 'please wait' prompt
while the 'main line' code shipped off the card details to the processor.
When I got the processor's response (which was usually before the prompt
finished playing), the 'main line' code would 'join' the prompt thread. If
the prompt thread was finished playing the prompt, execution continued
immediately. If the prompt thread was still playing the prompt, execution
continued as soon as the prompt thread received the AGI response from
Asterisk. None of this took any special programming or synchronizing from
me. It all 'just worked' because of the way pthreads are implemented.
So, what did I learn that would be of value to you? Really understand the
AGI protocol and you won't make a bunch of silly mistakes.
The AGI protocol is very simple:
1) Read the AGI environment from stdin before you do anything in your AGI
that interacts with Asterisk.
2) Write your AGI request to stdout.
3) Read Asterisk's AGI response from stdin.
4) Rinse, lather, repeat (steps 2 3).
If you use an established AGI library (like you really, really should),
this should be taken care of automagically, but understanding what's really
going on will help when things don't work as expected.
The number 1 mistake new AGI programmers make is not reading the AGI
environment. Sometimes they 'get away' with it, some times they don't.
The number 2 mistake is not reading the responses (step 3). Again,
sometimes it works, sometimes it doesn't
The number 3 mistake new (and old salts like me still make on occasion) is
doing any I/O on stdin or stdout. A lot of bad debugging centers around
'throw in a printf somewhere to see what's going on' instead of using 'real
programmer' tools like gdb*.
If you're using an established AGI library, the crucial relationship
between steps 2 and 3 will be take care of -- unless you're using multiple
threads.
If thread 1 issues an AGI request (writing to stdout) and then thread 2
issues an AGI request (also writing to stdout) before thread 1 reads it's
response, you've broken the protocol and bad things will happen. Maybe not
immediately, but certainly when you're demonstrating the system to your
boss.
So my first suggestion (after using an established library and without
knowing what your use case is) would be to 'designate' a single thread as
'the Asterisk' thread and only interact (via stdin and stdout) with
Asterisk in that thread. If that's not feasible, you're going to need some
sort of 'semaphore' mechanism so you don't 'intermingle' your AGI requests
and responses and thereby violate the AGI protocol.
Another couple of suggestions unrelated
