Re: [asterisk-users] Monitoring connection to VoIP provider?
On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com wrote: Community can help you better if you provide some details about you scenario and requirement. It's a very simple scenario: The Asterisk server is connected to a VoIP provider for calls to the PSTN, and I'd like to have Asterisk (or some other app) monitor the connection so that I can tell how good it is at any time, especially before calling out or receiving a call. The VoIP provides doesn't support any tool, eg. iperf. Is tracert/ping the only tools available in that scenario? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SoftHangup on asterisk 1.8.3.2 (renamed)
On Thu, 2011-07-07 at 14:23 -0400, Jeremy Kister wrote:
On 7/7/2011 9:32 AM, Ishfaq Malik wrote:
I'm having the same issue on 1.8.3.2 (with a couple of patches)
exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
This turned out to be a PEBKAC error. A newline was attached to the
$CHAN variable.
adding | tr -d '\n' to the end of the command fixed it right up.
Well in that case I'm having a different issue.
When I do
channel request hangup SIP/-1136
I get a
Requested Hangup on channel 'SIP/-1136'
response but the channel never hangs up
I'm having to restart the asterisk to clear the channels and that is not
an optimum solution!
Has anyone else encountered this or can see something obvious that I'm
doing wrong?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
On Mon, Jul 11, 2011 at 02:29:25PM -0700, Steve Edwards wrote: The second AGI, 'neutered-agi' is an AGI of 'production length' (around 1,600 lines) and supporting access to a MySQL database. The AGI is of 'production length' but still exits after reading the AGI environment variables because we are measuring program startup time. Perl's startup time also depends on the modules you load: The following is a simple loop that runs a perl program doing nothing (executing an empty statement) but firt loading a different set of modules for the task. The loop repeats it 1000 times: # No extra modules: $ time for i in `seq 1000`; do perl -e ''; done real0m5.093s user0m1.676s sys 0m2.524s # Asterisk::AGI: $ time for i in `seq 1000`; do perl -MAsterisk::AGI -e ''; done real0m9.400s user0m6.368s sys 0m2.060s # MySQL access: $ time for i in `seq 1000`; do perl -MDBI -MDBD::mysql -e ''; done real0m40.964s user0m33.122s sys 0m5.416s # Asterisk::AGI and MySQL access: $ time for i in `seq 1000`; do perl -MAsterisk::AGI -MDBI -MDBD::mysql -e ''; done real0m45.898s user0m36.798s sys 0m6.048s -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Monitoring connection to VoIP provider?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Tuesday, July 12, 2011 3:42 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Monitoring connection to VoIP provider? On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com wrote: Community can help you better if you provide some details about you scenario and requirement. It's a very simple scenario: The Asterisk server is connected to a VoIP provider for calls to the PSTN, and I'd like to have Asterisk (or some other app) monitor the connection so that I can tell how good it is at any time, especially before calling out or receiving a call. The VoIP provides doesn't support any tool, eg. iperf. Is tracert/ping the only tools available in that scenario? qualify=yes in the sip.conf entry for your provider. sip show peers in the Asterisk CLI to see the latency for each sip.conf entry. -- _ -- 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] FXO ports locking up
Doesn't seem to help. I did it early yesterday morning and have another 'stuck' call this morning Does anyone have any other ideas on what I can do to correct this? thanks Shawn CLI core show channels Channel Location State Application(Data) DAHDI/8-1(None) Up AppDial((Outgoing Line)) SIP/cordless8-04 725@out-phone8:1 Up Dial(DAHDI/8/725) 2 active channels 1 active call CLI core show channel DAHDI/8-1 -- General -- Name: DAHDI/8-1 Type: DAHDI UniqueID: 1310421996.2359 Caller ID: 725 Caller ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x4 (ulaw) ReadFormat: 0x4 (ulaw) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 23 Frames in: 2489590 Frames out: 72966 Time to Hangup: 0 Elapsed Time: 13h49m51s Direct Bridge: SIP/cordless8-049c Indirect Bridge: SIP/cordless8-049c -- PBX -- Context: in-phone8 Extension: Priority: 1 Call Group: 0 Pickup Group: 0 Application: AppDial Data: (Outgoing Line) Blocking in: ast_waitfor_nandfds Variables: BRIDGEPVTCALLID=2e52745c-7bdfef53@192.168.0.134 BRIDGEPEER=SIP/cordless8-049c DIALEDPEERNUMBER=8/725 TRANSFERCAPABILITY=SPEECH On Fri, Jul 8, 2011 at 7:25 PM, Alec Davis siva...@paradise.net.nz wrote: Is there a way to detect that there is no longer really an active call happening and force a hangup or reset the channel? It'd be great if this could happen automatically. Or as a temporary fix , is there a way to setup and extension that the SIP phone could dial which would clear any active calls associated with it? Right now if this happens, I need to login to the Asterisk CLI and issue a hangup command. If I don't, the channel appears to be in-use forever. This may be the answer sip.conf: ;--- RTP timers ; These timers are currently used for both audio and video streams. The RTP timeouts ; are only applied to the audio channel. ; The settings are settable in the global section as well as per device ; rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity ; on the audio channel ; when we're not on hold. This is to be able to hangup ; a call in the case of a phone disappearing from the net, ; like a powerloss or grandma tripping over a cable. Alec Davis -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO ports locking up
On Fri, Jul 08, 2011 at 10:58:06AM -0400, Shawn L wrote: I have a situation where I have an Asterisk box which receives 8 analog lines from a Mitel PBX and then drives 8 cordless SIP phones in a 1-to-1 mapping (a call coming in on port 1 of the digium FXO board is delivered to SIP phone 1, an outgoing call on SIP phone 2 goes out FXO line 2, etc. This works fine normally, but every once in a while (no set time, or pattern that I can see -- It may be caused by the wifi sip phone going out of range of an access point and not coming back into range fast enough) the FXO port does not hangup after the call is terminated and just sits in an in-use state. Since it's a 1-to-1 mapping, the SIP phone associated with the in-use line now produces a fast busy when you attempt to make a call because it cannot get an outbound line. Is there a way to detect that there is no longer really an active call happening and force a hangup or reset the channel? It'd be great if this could happen automatically. Or as a temporary fix , is there a way to setup and extension that the SIP phone could dial which would clear any active calls associated with it? Right now if this happens, I need to login to the Asterisk CLI and issue a hangup command. If I don't, the channel appears to be in-use forever. look for 'busydetect' in chan_dahdi.conf . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] DB Driven IVR
* DHAVAL , can you help me in designing the same ?* On Sat, Jul 9, 2011 at 5:50 PM, G M gm.cu...@gmail.com wrote: Anyone has Experience ? On Fri, Jul 8, 2011 at 2:18 PM, G M gm.cu...@gmail.com wrote: I am using Vicidial and I am looking for someone who can help with DB Driven IVR. -- _ -- 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] skype for asterisk usage in the future
On 07/11/2011 09:48 PM, d tbsky wrote: 1. SFA can not be registered after 26 July. so I want to prepare a backup machine for our server. I read in the document that I can re-register my SFA once. so I want to make sure if I can re-register with my backup server now, and in the same time my production machine still function correctly. and if my production machine is broken one day, my backup machine can go on line. in short: can I keep two machines with one license now? I hope so because I can not re-register later after 26 July. Yes, this is acceptable. 2. I saw SFA will not be supported after two years. my question is: although it is not supported, can I still use it? I want to buy more licenses now if I can still use it after two years even without official support. It is unknown whether it will continue to be usable after that period; Skype has the ability to disable SFA from accessing the Skype network if they feel that is what they want to do. Since it won't get any updates between now and then, it is very likely to be obsolete (from a 'Skype protocol' point of view) in two years and it seems quite likely that they won't want it accessing the network any more. It would be best to plan for it being non-functional after the two year support period is over. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] skype for asterisk usage in the future
On 07/12/2011 08:26 AM, Kevin P. Fleming wrote: It is unknown whether it will continue to be usable after that period; Skype has the ability to disable SFA from accessing the Skype network if they feel that is what they want to do. Since it won't get any updates between now and then, it is very likely to be obsolete (from a 'Skype protocol' point of view) in two years and it seems quite likely that they won't want it accessing the network any more. It would be best to plan for it being non-functional after the two year support period is over. Is there a project to replace Skype with a free software? Bob Rawlinson -- _ -- 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] ${HASH(SIP_CAUSE, ...)} and peer name
On Tue, Jul 12, 2011 at 00:22, Philippe Sultan philippe.sul...@gmail.comwrote:
The destination channel dies right after your Dial statement exits,
but you can retrieve the info in the channel that's still alive :
exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code :
${HASH(SIP_CAUSE,${CDR(dstchannel)})})
Works fine on the Asterisk server I'm running (1.8.3.3).
Thanks, that works for me as well.
Philippe
Ido
On Mon, Jul 11, 2011 at 11:01 PM, ik ido...@gmail.com wrote:
Hello,
I'm trying to figure out what was the return code of SIP for a call.
The problem is that HASH(SIP_CAUSE) require a peer name, but when I try
to
retrieve the peer name using ${CHANNEL(peername)}, I have an error
message
that CHANNEL does not have peername or it is not available to be used.
I tried to print it with NOOP on a live channel, and also after hangup,
both
with the same error message.
So how can I get SIP_CAUSE, or how can I get the peer name ?
Thanks,
Ido
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Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
Steve Edwards wrote:
Also they tend to be used more by 'non-programmers' who get away with
'stupid' stuff like calling out to system() and piping a bunch of
commands together because they don't know how to use the language
properly :)
I'm not disparaging Perl programmers or the language. I'm just saying
it is easier to 'abuse' a language like Perl or PHP than C.
This Bash snippet was posted to the -users list a couple of years ago.
I'm not trying to embarrass the original programmer or trash his
skills, I'm just using this snippet as an example. We've all got
skeletons in our closets :)
while read line; do
epoch=`echo $line | cut -d '|' -f 1`
if [ $epoch -ge $start_epoch -a $epoch -le $end_epoch ]; then
echo $line
fi
done /var/log/asterisk/queue_log
Note the second line. It creates a couple (2 or 3) of processes to
extract the first field from the queue_log file. For every line in
the input file!
This kind of coding is easy in a scripting language. It would be way
more difficult in C. So much more difficult, a programmer with the
requisite skills to do it should recognize the inefficiency and do
it another way.
If the original programmer had a better grasp of the language, he
could have coded this line as:
epoch=${line:0:10}
Since the Epoch will be 10 digits for the next 300 years, I'd feel
relatively comfortable with this solution.
This single change reduced the execution time of his script by an
order of magnitude.
Recoding it in a language more appropriate to processing lots of data
(like C) would reduce the execution time to 1/3,000th of the original. And
yes, I did it and measured it.
A skilled programmer (like any craftsman) has many tools in his toolbox,
the experience to choose the right one, and the skill to use it well.
C is my sharpest tool so I tend to see everything through that lens, but
I'm learning to appreciate PHP and how it lets me represent some
programming problems clearly, quickly and sufficiently efficient.
Steve,
I recognized the code you posted. It's mine:
http://lists.digium.com/pipermail/asterisk-users/2009-September/237750.html
Thank goodness you didn't try to embarrass me. You just used my code as an
example of how a non-programmer would use a language, called piping commands
together stupid and an indication of not knowing how to use the language,
referred to it as a skeleton in my closet and something that a programmer
with the requisites skills and a better grasp of the language wouldn't do.
I guess I am just not a skilled enough programmer to try to help someone out
on this list without getting criticized by you at the time
http://lists.digium.com/pipermail/asterisk-users/2009-September/237755.html
and again almost two years later.
Maybe, just maybe, it's possible that some people recognize that your crusade
to replace all scripting with compiled C has its virtues, but it's not worth
their programming time to save a few seconds of execution time. You've been
at this for years and yet scripting languages still exist.
Just think how fast Linux would boot if all of the init scripts were
rewritten in C and compiled (they probably have some pipes that could be
removed, too!!). Of course, it's pretty nice to be able to easily read and
modify them, but execution time is all that's important, right?
Your example reduced the execution time of my script by a factor of 2,700!!
That's great, but do you really think if I posted C source code as a response
(I could if I wanted to) to the original poster that she would have had any
idea what to do with it? I'd wager that she'd spend more than the minute and
a half your optimization saved pondering it before responding, What do I do
with this? Instead, she copy-and-pasted my code, ran it, said thanks (can
you believe she didn't even mention that pipe I used!!), and went on to the
next thing. So which is really more efficient?
Not to mention that your optimization is technically incorrect because it
assumes the field is a fixed length. I'll admit, it's a valid assumption,
but we're arguing technicalities here. I wonder if you'll be able to admit
that sometimes optimizing a one-off script just isn't worth it?
My code also has the benefit of being self-documenting. Anyone can look at
it and understand exactly what it does (quite useful for a mailing list,
don't you think?). On the other hand, most people will be headed straight
to the Advanced Bash Scripting Guide when they see epoch=${line:0:10}.
I haven't posted here in a long time and I usually make it a policy not to
get involved in or start any flame wars. Furthermore, I've always thought
that your posts are intelligent and show a real understanding of the subject
matter and a technical prowess. But you're out of line using other people's
code as examples of bad programming techniques. Unless my code is in a
directory on your desktop labeled Bad (but in NO Way
Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
Let's make this a Spider-man contest. The No-prize will be the satisfaction of seeing how it actually works. Write stevestest.agi in Perl, PHP and C. The program must load the AGI variables, do a MYSQL QUERY and a MYSQL INSERT. Post your source and results using this methodology: time for i in `seq 1000`; do ./stevestest.out ; done - C time for i in `seq 1000`; do php stevesetest.php '; done - PHP time for i in `seq 1000`; do perl stevestest.pl ; done Any takers? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Monday, July 11, 2011 5:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl On 12/07/11 9:29 AM, Steve Edwards wrote: Many times, I've made the statement that you can execute hundreds of AGIs written in C in the time it takes to load an interpreter and parse a script written in PHP or Perl. It would be interesting to see the same types of tests run against fast-agi - personally if I write an agi that will be called 1000 times I'm going to leave it running and have network requests against it rather than starting and stopping every time. Interesting tests nonetheless - I would have been pretty concerned if straight C wasn't faster :-) Mind if I post it to the Daily Asterisk News? -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/cc.php (Call Centre Solutions) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
On 07/12/2011 09:33 AM, Matthew J. Roth wrote: Just think how fast Linux would boot if all of the init scripts were rewritten in C and compiled (they probably have some pipes that could be removed, too!!). Of course, it's pretty nice to be able to easily read and modify them, but execution time is all that's important, right? OT: Take a look at 'systemd'; this is exactly what's happening there, and Fedora is likely to incorporate it into Fedora 16, and it will make its way into other distros after that. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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] Monitoring connection to VoIP provider?
A quick to implement open source network monitoring tool is smokeping: http://oss.oetiker.ch/smokeping/index.en.html Tobi Oetiker and Niko Tyni's awesome tool can monitor latency on a number of layers, and maintains charted records of connection quality. It has a probe specific to SIP: http://oss.oetiker.ch/smokeping/probe/SipSak.en.html While there are many great implementation guides out there, I've drafted a basic install (that is somewhat OS X centric) at: http://www.stevenstromer.com/grok/smokeping-installation-for-os-x-10-5-10-6 And a basic configuration guide: http://www.stevenstromer.com/grok/smokeping-configuration-for-a-home-or-small-business-network (it doesn't describe implementing the SIP probe just yet) Hope you find smokeping as helpful as I have, and not only for VoIP services. I am sure there are a number of other, more dedicated and equally useful apps. Also, I believe there have been numerous previous discussions in this thread about monitoring options. Look back. Steven On Thu, 7 Jul 2011 09:32:08 +0500, Faisal Hanif fai...@vopium.com wrote: Community can help you better if you provide some details about you scenario and requirement. It's a very simple scenario: The Asterisk server is connected to a VoIP provider for calls to the PSTN, and I'd like to have Asterisk (or some other app) monitor the connection so that I can tell how good it is at any time, especially before calling out or receiving a call. The VoIP provides doesn't support any tool, eg. iperf. Is tracert/ping the only tools available in that scenario? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] CDRs
Hi Like we can define cdr field format for csv, is it possible to define if cdrs are stored in a database? Also, what will be size limit for database CDR storage ? -- _ -- 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] CDRs
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Tuesday, July 12, 2011 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDRs Hi Like we can define cdr field format for csv, is it possible to define if cdrs are stored in a database? Also, what will be size limit for database CDR storage ? The code is on your machine. You can do whatever you want to with it. Caveat Emptor! -- _ -- 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] skype for asterisk usage in the future
On Tuesday 12 Jul 2011, d tbsky wrote: hi: I am a SFA (skype for asterisk) user. I had ask Digium questions about SFA usage in the future. but they seem too busy to reply. so I tried at this list. I hope there are SFA users or Digium people can solve my confusion. Poor you! To my mind, Skype with its opaque, proprietary protocols is the exact opposite of what telecommunications is supposed to be about. I can make a call, or send an SMS, from my HTC on Vodafone to my friend's Samsung on Tesco without thinking twice about it, and that's the way we all expect it to be. But if it hadn't been for governments enforcing standards, the mobile networks could well have ended up fragmentated; with different handset manufacturers and different network operators all using competing, proprietary standards to lock one another out and their customers in. 2. I saw SFA will not be supported after two years. my question is: although it is not supported, can I still use it? I want to buy more licenses now if I can still use it after two years even without official support. That depends entirely on whether Skype update their protocols and block out the old ones. Two years is easily long enough for them to do that; especially given the way Skype works. It could even do stealth upgrades in the background. Look at it this way: You've got two years to migrate away from Skype and start using something else -- and this time around, for the love of all that's sane and wholesome, be sure to choose something that supports open standards, so you can never get shafted the same way again. If someone manages successfully to reverse-engineer Skype during that time, you *might* have a little longer; but I wouldn't bet the family farm on that. -- AJS Answers come *after* questions. -- _ -- 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] Park/VoiceMail on DAHDI congestion
Hi Guys,
I have been trying to implement the following for days but with no success,
any help would be greatly appreciated
My asterisk box gets calls from the SIP interface and forwards to the DAHDI
interface for example
--sip.conf-
[smycontext]
type=friend
host=xxx
fromuser=xxx
context=mycontext
disallow=all
allow=g729
allow=g723
allow=ulaw
allow=alaw
--extensions.conf-
[mycontext]
exten = _0.,1,Dial(dahdi/g11/${EXTEN})
This works well, until the DAHDI channel gets filled up. and I get the error
[Jul 12 18:31:20] WARNING[7476]: app_dial.c:2041 dial_exec_full: Unable to
create channel of type 'dahdi' (cause 34 - Circuit/channel congestion)
[Jul 12 18:31:25] WARNING[7477]: app_dial.c:2041 dial_exec_full: Unable to
create channel of type 'dahdi' (cause 34 - Circuit/channel congestion)
Now what I want is:
1. When the DAHDI channel group is filled up, I want to park the call
somewhere (put it on hold and play some music or message)
2. When a DAHDI channel gets free connect the parked call by dial the
channel and bridge the two calls.
3. The parking should be first in first out
If this is impossible I want to send all calls to one voice mailbox, take a
message and hang up
I have not put what I have tried here because I don't want to bias the
reply's I'll get
Any help?
Thanks!
Chris
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Re: [asterisk-users] CDRs
Read the wiki / manuals From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of deeps backup Sent: Tuesday, July 12, 2011 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] CDRs Hi Like we can define cdr field format for csv, is it possible to define if cdrs are stored in a database? Also, what will be size limit for database CDR storage ? -- _ -- 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] Benchmarking AGI performance in C, PHP, and Perl
On Tue, Jul 12, 2011 at 10:06:12AM -0500, Kevin P. Fleming wrote: On 07/12/2011 09:33 AM, Matthew J. Roth wrote: Just think how fast Linux would boot if all of the init scripts were rewritten in C and compiled (they probably have some pipes that could be removed, too!!). Of course, it's pretty nice to be able to easily read and modify them, but execution time is all that's important, right? OT: Take a look at 'systemd'; this is exactly what's happening there, and Fedora is likely to incorporate it into Fedora 16, and it will make its way into other distros after that. Well, there are a number of separate optimizations in systemd: 1. Delayed loading of services (or even not loading them at all, if not needed. E.g.: don't load CUPS if nobody needs it. 2. Paralelized loading of services (though there have been other implemtnations of that. SuSE has had that for years: insserv). 3. Rewriting some scripts in C. Also note that udev has been going through this for quite some time as well. 4. A nice facility to get the timing information and show where the actual bottlenecks are. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] skype for asterisk usage in the future
hi: thanks for all the information. I don't use skype and I ban skype at our network. but there are some people who use skype and want us to use skype to contact them. SFA is my saver because our users can use their phone to talk with skype users and no need to install any skype software. I hope skype die asap. but if it is alive, I must find someway to satisfy these skype customers... 2011/7/12 A J Stiles asterisk_l...@earthshod.co.uk: On Tuesday 12 Jul 2011, d tbsky wrote: hi: I am a SFA (skype for asterisk) user. I had ask Digium questions about SFA usage in the future. but they seem too busy to reply. so I tried at this list. I hope there are SFA users or Digium people can solve my confusion. Poor you! To my mind, Skype with its opaque, proprietary protocols is the exact opposite of what telecommunications is supposed to be about. I can make a call, or send an SMS, from my HTC on Vodafone to my friend's Samsung on Tesco without thinking twice about it, and that's the way we all expect it to be. But if it hadn't been for governments enforcing standards, the mobile networks could well have ended up fragmentated; with different handset manufacturers and different network operators all using competing, proprietary standards to lock one another out and their customers in. 2. I saw SFA will not be supported after two years. my question is: although it is not supported, can I still use it? I want to buy more licenses now if I can still use it after two years even without official support. That depends entirely on whether Skype update their protocols and block out the old ones. Two years is easily long enough for them to do that; especially given the way Skype works. It could even do stealth upgrades in the background. Look at it this way: You've got two years to migrate away from Skype and start using something else -- and this time around, for the love of all that's sane and wholesome, be sure to choose something that supports open standards, so you can never get shafted the same way again. If someone manages successfully to reverse-engineer Skype during that time, you *might* have a little longer; but I wouldn't bet the family farm on that. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
On Tue, 12 Jul 2011, Matthew J. Roth wrote: Just think how fast Linux would boot if all of the init scripts were rewritten in C and compiled (they probably have some pipes that could be removed, too!!). Of course, it's pretty nice to be able to easily read and modify them, but execution time is all that's important, right? Very probably the wrong place to duscuss this, however... I build embedded linux boxes for a variety of purposes (including asterisk), and I have to say that boot-time isn't always a factor. Really, when you have a box with an uptime of: # uptime 22:22:21 up 1374 days, 2:42, 2 users, load average: 0.00, 0.02, 0.00 who really cares how long it take to boot. The Linux kernel itself can boot in under a second - trouble is, it can take 15-20 seconds going through a PCs BIOS to get there, then it's init scripts, loading modules, etc. A lot of these are waiting on hardware, or the network coming up and compiling them isn't going to help much... In properly embedded systems you can get to application very fast (no init/scripts!), however in more general purpose systems, I feel it's diminishing returns time. I've also moved from sysv-rc to file-rc too - which completely serialises scripts - which might seem a backwards step, but it gives easier control and I'm happy that my PBXs go from cold to ready in under a minute, and my (getting old now) Acer Aspire One goes from cold to workable GUI in 45 seconds. I can live with that. Gordon -- _ -- 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] Benchmarking AGI performance in C, PHP, and Perl
Tzafrir Cohen wrote: Well, there are a number of separate optimizations in systemd: 1. Delayed loading of services (or even not loading them at all, if not needed. E.g.: don't load CUPS if nobody needs it. 2. Paralelized loading of services (though there have been other implemtnations of that. SuSE has had that for years: insserv). 3. Rewriting some scripts in C. Also note that udev has been going through this for quite some time as well. 4. A nice facility to get the timing information and show where the actual bottlenecks are. Tzafrir, I kind of suspected that the init scripts weren't a great example because of systemd. It's already in Fedora 15, but I have to plead ignorance to its implementation. I thought it still used text-based service files, but I'm only familiar with it from reading the large number of posts it has generated on the Fedora users list. That's actually started to settle down and Gnome 3 has emerged as the new whipping boy. Since systemd has been proposed as a dependency of Gnome 3, this has the potential to create singularity that will destroy the universe. ; ) Personally, I'm not really excited about any of the optimizations that you listed. I can manage which services are started and the Linux boot sequence is pretty fast already. It's far quicker than the serial initialization of hardware that our servers go through at each boot. So for my use case, it can still be cited as an example of optimizing the wrong thing. In the long run, it's likely to save me very little time compared to the amount of time spent learning the new system. I'm sure it has its benefits and I'll come to appreciate them over time, but SysV init ain't broke (for me) so I'm not looking forward to fixing it. 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] REALY strange issue with making calls biside 2 phones
Thats my issue, i hope someone could suggest something: Phone A - Phone B == Using SIP RTP CoS mark 5 -- Executing [01@default:1] Dial(SIP/00-0076, SIP/01) in new stack == Using SIP RTP CoS mark 5 -- Called 01 -- SIP/01-0077 is ringing -- SIP/01-0077 answered SIP/00-0076 -- Locally bridging SIP/00-0076 and SIP/01-0077 == Spawn extension (default, 01, 1) exited non-zero on 'SIP/00-0076' Phone B - phone A == Using SIP RTP CoS mark 5 -- Executing [00@default:1] Dial(SIP/01-0078, SIP/00) in new stack [Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [00@default:2] Hangup(SIP/01-0078, ) in new stack == Spawn extension (default, 00, 2) exited non-zero on 'SIP/01-0078' -- -- Best regards Matiss Jekabsons Procerto Ltd. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dtmf issues solved
Deployed a new server different mobo and problem went away. Same version of asterisk, same sangoma card. Sent from my android device.-- _ -- 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] Mysterious dropped calls
So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious dropped calls. This only happens on calls that are outbound on Dahdi and mostly happens in conference calls particularly 8xx-xxx- This is the output of the hangup. [Ksebpbx1*CLI [0KPRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request, hold-state Idle PRI Span: 1 q931.c:4845 q931_disconnect: Call 32985 enters state 11 (Disconnect Request). Hold state: Idle PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: DISCONNECT (69) PRI Span: 1 TEI=0 Transmitting N(S)=51, window is open V(A)=51 K=7 PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: DISCONNECT (69) PRI Span: 1 [08 02 81 90] PRI Span: 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) PRI Span: 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'DAHDI/i1/18662006965-1be' [Ksebpbx1*CLI [0K == Spawn extension (from-sip, 18662006965, 1) exited non-zero on 'SIP/7027-0520' [Ksebpbx1*CLI [0KPRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=5 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent to originator) [Ksebpbx1*CLI [0KPRI Span: 1 Message Type: RELEASE (77) PRI Span: 1 Received message for call 0x20984f0 on 0x7f7f804f24d0 TEI/SAPI 0/0, call-pri is 0x7f7f804f24d0 TEI/SAPI 0/0 PRI Span: 1 q931.c:7237 post_handle_q931_message: Call 32985 enters state 0 (Null). Hold state: Idle Span: 1 Processing event: PRI_EVENT_HANGUP PRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request, hold-state Idle PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: RELEASE COMPLETE (90) PRI Span: 1 TEI=0 Transmitting N(S)=52, window is open V(A)=52 K=7 PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: RELEASE COMPLETE (90) PRI Span: 1 [08 02 81 90] PRI Span: 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) PRI Span: 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] PRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle PRI Span: 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle Any ideas? -- _ -- 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] Benchmarking AGI performance in C, PHP, and Perl
On Tue, 12 Jul 2011, Matthew J. Roth wrote:
I recognized the code you posted. It's mine:
Thank goodness you didn't try to embarrass me.
Thank you for acknowledging that it was not my intent.
You just used my code as an example of how a non-programmer would use
a language, called piping commands together stupid and an indication
of not knowing how to use the language, referred to it as a skeleton in
my closet and something that a programmer with the requisites skills and
a better grasp of the language wouldn't do.
Which part is not true? I did not say you were stupid, just that 1 line of
code is. Would you be more comfortable with 'ignorant?' After reading the
definitions at www.dictionary.com I think it may be more accurate and less
prone to misinterpretation but I don't know how to say a line of code is
ignorant.
I think you're being overly sensitive here.
I guess I am just not a skilled enough programmer to try to help someone
out on this list without getting criticized by you at the time
http://lists.digium.com/pipermail/asterisk-users/2009-September/237755.html
and again almost two years later.
A man that has caught his first fish can teach the man with an empty net.
You've been harboring this perceived offense in silence for 2 years? I've
re-read that post several times today and cannot see where I criticized
you or your code. Please point it out for me.
Maybe, just maybe, it's possible that some people recognize that your
crusade to replace all scripting with compiled C has its virtues, but
it's not worth their programming time to save a few seconds of execution
time. You've been at this for years and yet scripting languages still
exist.
Both compiled and scripting languages have their place. I have a box
wrench and a ratchet socket even though they both do kind of the same
thing.
I do a fair bit of scripting, just not on tasks that are repeated 500,000
times a day.
Just think how fast Linux would boot if all of the init scripts were
rewritten in C and compiled (they probably have some pipes that could be
removed, too!!). Of course, it's pretty nice to be able to easily read
and modify them, but execution time is all that's important, right?
If I rebooted my box 500,000 times a day, that may be relevant. My box
does execute 500,000 AGIs a day.
If I was playing the same MP3 500,000 times a day and complaining about
the performance might you suggest I could (or should) transcode the file
to SLIN, ULAW, PCM, etc?
On the other hand, most people will be headed straight to the Advanced
Bash Scripting Guide when they see epoch=${line:0:10}.
I consider that a win-win. I have fed them for the day and inspired them
to learn to fish better. Two birds, one stone in my book.
I replied to your code post 2 years ago because you warned the original
recipient that 'it may take a while to complete.'
To me, that's a call to action for a programmer or any kind of engineer.
Part of the reason I feel compelled to post 'improvements' is to 'pay it
forward' for the programmers that help me learn my craft. Also, if I let
it be, future readers will happily (ignorantly?) implement this construct
for eternity.
I'm reminded of one of my favorite jokes.
A husband is observing his wife is preparing their first holiday meal. She
takes the ham out of the fridge, sets it on the block, hacks off the last
4 inches and tosses the piece in the trash.
The husband, cognizant of his ignorance in 'the ways of the kitchen' asks
her why she did this.
She stops, looks quizzically at the ceiling and says I don't know. I do
it because my mom always did it.
The husband calls her mom. Her explanation is the same 'I don't know. I do
it because my mom always did it.
The husband calls the grandmother. Her explanation is 'I don't know why
they do that. I do it because my oven is too small.'
Almost a year ago, Anthony Messina posted a link to his fax gateway script
(now at http://messinet.com/trac/wiki/AsteriskFAXGateway). My hubris
deluded me to think I could make it better. I got 'skooled!'
I made a list of about 20 constructs I had no clue what they did and
headed out to the 'Advanced Bash-Scripting Guide'
(http://tldp.org/LDP/abs/html/)
I've incorporated some of what I learned and my Bash scripting is much
better for it.
I always try to read Tzafrir Cohen's posts because it's obvious he knows
way more than I do about Unix at a systems level.
Ditto for (in alphabetical order) Gordon Henderson, Kevin P. Fleming,
Steve Underwood, Tilghman Lesher, Tony Mountifield and many others. We sit
at the feet of giants.
But you're out of line using other people's code as examples of bad
programming techniques. Unless my code is in a directory on your desktop
labeled Bad (but in NO Way Embarrassing) Programming Examples it would
have been more efficient (see what I did there?) to come up with
something on your own.
My sincere apologies. I did not
Re: [asterisk-users] Mysterious dropped calls
So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious dropped calls. This only happens on calls that are outbound on Dahdi and mostly happens in conference calls particularly 8xx-xxx- This is the output of the hangup. [Ksebpbx1*CLI [0KPRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request, hold-state Idle PRI Span: 1 q931.c:4845 q931_disconnect: Call 32985 enters state 11 (Disconnect Request). Hold state: Idle PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: DISCONNECT (69) PRI Span: 1 TEI=0 Transmitting N(S)=51, window is open V(A)=51 K=7 PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: DISCONNECT (69) PRI Span: 1 [08 02 81 90] PRI Span: 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) PRI Span: 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'DAHDI/i1/18662006965-1be' [Ksebpbx1*CLI [0K == Spawn extension (from-sip, 18662006965, 1) exited non-zero on 'SIP/7027-0520' [Ksebpbx1*CLI [0KPRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=5 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent to originator) [Ksebpbx1*CLI [0KPRI Span: 1 Message Type: RELEASE (77) PRI Span: 1 Received message for call 0x20984f0 on 0x7f7f804f24d0 TEI/SAPI 0/0, call-pri is 0x7f7f804f24d0 TEI/SAPI 0/0 PRI Span: 1 q931.c:7237 post_handle_q931_message: Call 32985 enters state 0 (Null). Hold state: Idle Span: 1 Processing event: PRI_EVENT_HANGUP PRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request, hold-state Idle PRI Span: 1 PRI Span: 1 DL-DATA request PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: RELEASE COMPLETE (90) PRI Span: 1 TEI=0 Transmitting N(S)=52, window is open V(A)=52 K=7 PRI Span: 1 PRI Span: 1 Protocol Discriminator: Q.931 (8) len=9 PRI Span: 1 TEI=0 Call Ref: len= 2 (reference 217/0xD9) (Sent from originator) PRI Span: 1 Message Type: RELEASE COMPLETE (90) PRI Span: 1 [08 02 81 90] PRI Span: 1 Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) PRI Span: 1 Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] PRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle PRI Span: 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Idle Any ideas? The decision to drop the call within Asterisk has already been made and is not shown in the trace. This trace is just showing the clearing of the call being initiated by the Asterisk side with a cause of normal clearing. Nothing is unexpected here. You need to capture debug output of earlier events to figure this out. Richard -- _ -- 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] Mysterious dropped calls
Sent from a computer -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Rosedale Sent: Tuesday, July 12, 2011 4:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mysterious dropped calls So I'm now using asterisk 1.8.5rc1 for Asterisk. I'm still getting mysterious dropped calls. This only happens on calls that are outbound on Dahdi and mostly happens in conference calls particularly 8xx-xxx- This is the output of the hangup. [Ksebpbx1*CLI [0KPRI Span: 1 q931_hangup: other hangup PRI Span: 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, busydetect=yes or callprogress=yes in chan_dahdi.conf often cause random call hangups. If you have those options set, either remove them or set them to no. -- _ -- 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] Benchmarking AGI performance in C, PHP, and Perl
Steve, Apology accepted. As I said in the original post, I hold you in high regard so your criticism was hard to take. I still think that the trade- off between readability and optimization is up for debate, but it's certainly nothing to hold a grudge over. I can tell you one thing for certain: I will think of you and that Bash substring-extraction construct every time I even consider piping something to cut. I might even remember that particular syntax without resorting to the ABS. I'm not making any promises about the others, though. Some of them are just nasty. ; ) http://tldp.org/LDP/abs/html/refcards.html#AEN22429 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
Re: [asterisk-users] REALY strange issue with making calls biside 2 phones
what does sip show peers say? On Tue, Jul 12, 2011 at 12:40 PM, Matiss Jekabsons mat...@jekabsons.lv wrote: Thats my issue, i hope someone could suggest something: Phone A - Phone B == Using SIP RTP CoS mark 5 -- Executing [01@default:1] Dial(SIP/00-0076, SIP/01) in new stack == Using SIP RTP CoS mark 5 -- Called 01 -- SIP/01-0077 is ringing -- SIP/01-0077 answered SIP/00-0076 -- Locally bridging SIP/00-0076 and SIP/01-0077 == Spawn extension (default, 01, 1) exited non-zero on 'SIP/00-0076' Phone B - phone A == Using SIP RTP CoS mark 5 -- Executing [00@default:1] Dial(SIP/01-0078, SIP/00) in new stack [Jul 12 19:08:35] WARNING[2965]: app_dial.c:2039 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [00@default:2] Hangup(SIP/01-0078, ) in new stack == Spawn extension (default, 00, 2) exited non-zero on 'SIP/01-0078' -- -- Best regards Matiss Jekabsons Procerto Ltd. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
On Mon, Jul 11, 2011 at 06:45:08PM -0700, Steve Edwards wrote:
Also they tend to be used more by 'non-programmers' who get away with
'stupid' stuff like calling out to system() and piping a bunch of
commands together because they don't know how to use the language
properly :)
On Mon, 11 Jul 2011, cbul...@gmail.com wrote:
I understand your point but I don't share it There are a lot
Asterisk-Perl project working in production environment.
Then I didn't communicate my point clearly.
I'm not disparaging Perl programmers or the language. I'm just saying it
is easier to 'abuse' a language like Perl or PHP than C.
This Bash snippet was posted to the -users list a couple of years ago.
I'm not trying to embarrass the original programmer or trash his skills,
I'm just using this snippet as an example. We've all got skeletons in our
closets :)
while read line; do
epoch=`echo $line | cut -d '|' -f 1`
if [ $epoch -ge $start_epoch -a $epoch -le $end_epoch ]; then
echo $line
fi
done /var/log/asterisk/queue_log
Note the second line. It creates a couple (2 or 3) of processes to
extract the first field from the queue_log file. For every line in the
input file!
This kind of coding is easy in a scripting language. It would be way more
difficult in C. So much more difficult, a programmer with the requisite
skills to do it should recognize the inefficiency and do it another way.
If the original programmer had a better grasp of the language, he could
have coded this line as:
epoch=${line:0:10}
Since the Epoch will be 10 digits for the next 300 years, I'd feel
relatively comfortable with this solution.
For the record, I suspect you meant:
epoch=${line%%|*}
This actually does exactly what is written on the above command (without
having to assume the length of the field), and also does not use
bash-specific syntax and hence can be used with a faster shell like
dash (which Debian and Ubuntu have as /bin/sh by default).
This single change reduced the execution time of his script by an order
of magnitude.
Recoding it in a language more appropriate to processing lots of data
(like C) would reduce the execution time to 1/3,000th of the original.
And yes, I did it and measured it.
Actually, that mistake is easy to make in shell scripts. But even simple
script languages such as awk fare much better here:
awk -F'|' {
epoch=\$1;
if (epoch = $start_epoch epoch = $end_epoch) {
print epoch
}
} /var/log/asterisk/queue_log
How much time will it take you to write something as fast as this in C?
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.co...@xorcom.com
+972-50-7952406 mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir
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Re: [asterisk-users] Benchmarking AGI performance in C, PHP, and Perl
On Mon, Jul 11, 2011 at 06:45:08PM -0700, Steve Edwards wrote:
while read line; do
epoch=`echo $line | cut -d '|' -f 1`
if [ $epoch -ge $start_epoch -a $epoch -le $end_epoch ]; then
echo $line
fi
done /var/log/asterisk/queue_log
[snipping snippy comments about improving the second line]
epoch=${line:0:10}
Since the Epoch will be 10 digits for the next 300 years, I'd feel
relatively comfortable with this solution.
On Wed, 13 Jul 2011, Tzafrir Cohen wrote:
For the record, I suspect you meant:
epoch=${line%%|*}
I didn't mean it, but I like it. Thanks.
Actually, that mistake is easy to make in shell scripts. But even simple
script languages such as awk fare much better here:
awk -F'|' {
epoch=\$1;
if (epoch = $start_epoch epoch = $end_epoch) {
print epoch
}
} /var/log/asterisk/queue_log
1) The 2 conditionals should be swapped.
2) I think you meant 'print \$0' instead of 'print epoch'
We'll leave defining start_epoch and end_epoch as an exercise for the
reader :)
How much time will it take you to write something as fast as this in C?
Well, the C version was still 10 times faster, but I concede it took a
whole lot longer to write.
The question always comes down to how many times are you going to use it
and how much will it cost to make it faster.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Problem on Dialling-out
Hi List, I have a Asterisk + FreePbx Server setup with around 10 SIP extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is being dropped with the following message on asterisk log: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set(SIP/1001-0014, RC=0) in new stack -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(SIP/1001-0014, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,1) -- Executing [continue@macro-dialout-trunk:1] GotoIf(SIP/1001-0014, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,3) -- Executing [continue@macro-dialout-trunk:3] NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:4] Set(SIP/1001-0014, CALLERID(number)=1001) in new stack -- Executing [639285010430@from-internal:8] Macro(SIP/1001-0014, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, ) in new stack -- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, all-circuits-busy-now,noanswer) in new stack -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') -- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, pls-try-call-later,noanswer) in new stack -- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en') -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1001-0014' in macro 'outisbusy' == Spawn extension (from-internal, 639285010430, 8) exited non-zero on 'SIP/1001-0014' -- Executing [h@from-internal:1] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0014' localhost*CLI Can someone assist me please. Thanks in advance. Regards, Malvin -- _ -- 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 on Dialling-out
Your trunk shows busy: * -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)* Try this in the CLI (asterisk -r): *core set verbose 0* *sip set debug peer CordiaVoIP* And then make a call and read why the SIP trunk is failing. -Bruce On Wed, Jul 13, 2011 at 12:23 AM, Malvin Rito mr...@mail.altcladding.com.ph wrote: Hi List, I have a Asterisk + FreePbx Server setup with around 10 SIP extensions and 1 VoIP trunk (CordiaVoIP), when we dial-out to any number call is being dropped with the following message on asterisk log: == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called CordiaVoIP/639285010430 -- SIP/CordiaVoIP-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [s@macro-dialout-trunk:20] NoOp(SIP/1001-0014, Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0) in new stack -- Executing [s@macro-dialout-trunk:21] Goto(SIP/1001-0014, s-CONGESTION,1) in new stack -- Goto (macro-dialout-trunk,s-**CONGESTION,1) -- Executing [s-CONGESTION@macro-dialout-**trunk:1] Set(SIP/1001-0014, RC=0) in new stack -- Executing [s-CONGESTION@macro-dialout-**trunk:2] Goto(SIP/1001-0014, 0,1) in new stack -- Goto (macro-dialout-trunk,0,1) -- Executing [0@macro-dialout-trunk:1] Goto(SIP/1001-0014, continue,1) in new stack -- Goto (macro-dialout-trunk,continue,**1) -- Executing [continue@macro-dialout-trunk:**1] GotoIf(SIP/1001-0014, 1?noreport) in new stack -- Goto (macro-dialout-trunk,continue,**3) -- Executing [continue@macro-dialout-trunk:**3] NoOp(SIP/1001-0014, TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks) in new stack -- Executing [continue@macro-dialout-trunk:**4] Set(SIP/1001-0014, CALLERID(number)=1001) in new stack -- Executing [639285010430@from-internal:8] Macro(SIP/1001-0014, outisbusy,) in new stack -- Executing [s@macro-outisbusy:1] Progress(SIP/1001-0014, ) in new stack -- Executing [s@macro-outisbusy:2] Playback(SIP/1001-0014, all-circuits-busy-now,**noanswer) in new stack -- SIP/1001-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') -- Executing [s@macro-outisbusy:3] Playback(SIP/1001-0014, pls-try-call-later,noanswer) in new stack -- SIP/1001-0014 Playing 'pls-try-call-later.gsm' (language 'en') -- Executing [s@macro-outisbusy:4] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/1001-0014' in macro 'outisbusy' == Spawn extension (from-internal, 639285010430, 8) exited non-zero on 'SIP/1001-0014' -- Executing [h@from-internal:1] Macro(SIP/1001-0014, hangupcall) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1001-0014, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,4) -- Executing [s@macro-hangupcall:4] GotoIf(SIP/1001-0014, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,7) -- Executing [s@macro-hangupcall:7] GotoIf(SIP/1001-0014, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [s@macro-hangupcall:9] Hangup(SIP/1001-0014, ) in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/1001-0014' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/1001-0014' localhost*CLI Can someone assist me please. Thanks in advance. Regards, Malvin -- __**__**_ -- 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:
Re: [asterisk-users] Problem on Dialling-out
Sorry I do not understand it, here is result after: Audio is at 172.16.9.15 port 15022 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 64.211.94.211:5060: INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0 Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport Max-Forwards: 70 From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc To: sip:639285010...@lasip1.cordiaip.net Contact: sip:Unknown@172.16.9.15 Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7 Date: Wed, 13 Jul 2011 04:07:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 45158429 45158429 IN IP4 172.16.9.15 s=Asterisk PBX 1.6.2.7 c=IN IP4 172.16.9.15 t=0 0 m=audio 15022 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (no NAT) to 64.211.94.211:5060: INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0 Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport Max-Forwards: 70 From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc To: sip:639285010...@lasip1.cordiaip.net Contact: sip:Unknown@172.16.9.15 Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7 Date: Wed, 13 Jul 2011 04:07:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 45158429 45158429 IN IP4 172.16.9.15 s=Asterisk PBX 1.6.2.7 c=IN IP4 172.16.9.15 t=0 0 m=audio 15022 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to 64.211.94.211:5060: INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0 Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport Max-Forwards: 70 From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc To: sip:639285010...@lasip1.cordiaip.net Contact: sip:Unknown@172.16.9.15 Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7 Date: Wed, 13 Jul 2011 04:07:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 45158429 45158429 IN IP4 172.16.9.15 s=Asterisk PBX 1.6.2.7 c=IN IP4 172.16.9.15 t=0 0 m=audio 15022 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #3 (no NAT) to 64.211.94.211:5060: INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0 Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport Max-Forwards: 70 From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc To: sip:639285010...@lasip1.cordiaip.net Contact: sip:Unknown@172.16.9.15 Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7 Date: Wed, 13 Jul 2011 04:07:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 45158429 45158429 IN IP4 172.16.9.15 s=Asterisk PBX 1.6.2.7 c=IN IP4 172.16.9.15 t=0 0 m=audio 15022 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #4 (no NAT) to 64.211.94.211:5060: INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0 Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport Max-Forwards: 70 From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc To: sip:639285010...@lasip1.cordiaip.net Contact: sip:Unknown@172.16.9.15 Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7 Date: Wed, 13 Jul 2011 04:07:34 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 281 v=0 o=root 45158429 45158429 IN IP4 172.16.9.15 s=Asterisk PBX 1.6.2.7 c=IN IP4 172.16.9.15 t=0 0 m=audio 15022 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #5 (no NAT) to 64.211.94.211:5060: INVITE sip:639285010...@lasip1.cordiaip.net SIP/2.0 Via: SIP/2.0/UDP 172.16.9.15:5060;branch=z9hG4bK68d45015;rport Max-Forwards: 70 From: Cordia sip:Unknown@172.16.9.15;tag=as2267fdcc To: sip:639285010...@lasip1.cordiaip.net Contact: sip:Unknown@172.16.9.15 Call-ID: 12d2279238e5851572c30cad11bb9492@172.16.9.15 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.7 Date: Wed, 13 Jul 2011
