[asterisk-users] Paltel subscribers as called parties for SIP attacks (was: Malicious traffic comming from 37.75.210.90)
For what it's worth, I see similar traffic regularly from: orange.ps hadara.ps ovh.net iweb.ca scalabledns.com securedservers.com wholesaleinternet.com hostnoc.net rackspace.com hetzner.de all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers). Common numbers are: 972592871970 972597562803 972592170729 972595936848 972599532957 972592170729 972592539831 972592910519 972592577022 972592648299 972599146173 972592264761 972592600109 972598285108 972592910519 972599463826 972597072204 972599327923 972595813485 972598642462 972598431470 972598372537 972597248231 972598431470 … Now some of these numbers have been short-lived, others have been in use more than 2 years, like 972597562803 which seems to be sloppy tradecraft. Why would an internet subscriber from hadara.ps, for instance, want to call a Paltel mobile user via some remotely hacked SIP PBX thousands of miles away given than Paltel is partially owned by Hadara Technology Investment Co. (and Paltel leases long-haul infrastructure from Hadara anyway)? http://en.wikipedia.org/wiki/Paltel Well, if the Paltel subscriber were actually abroad… say in the US or Algeria or the Philippines, but he didn't want to risk the longest arm of the call being intercepted by Echelon or similar means, then he'd find an ISP in the country which he knew that subscriber to currently be in, and scan its CIDR blocks for insecure SIP PBX's to use to contact the mobile user… relying on domestic privacy protections to inhibit spying on internal traffic to that country. Perhaps Hadara (or a Hamas cell operating within Hadara) has moved from psyops to more overt means: http://blogs.norman.com/2012/security-research/cyberattack-against-israeli-and-palestinian-targets-for-a-year I'm surprised that DHS hasn't taken more interest in this. Or perhaps they already have, and are operating deliberately insecure PBX's as honeypots. Coming soon to your AGPS+ coordinates: a Predator drone… In any case, with all the SIP (and other) abuse I've received from Hadara.ps, they've never once acknowledged a complaint I've sent in… which seems to be tacit approval of the practice. I'd be curious to know what everyone else's experiences have been like, and why 95% or better of the SIP attacks on my PBX are destined for Paltel mobile subscribers. Given the number of inhabitants in Gaza, it seems like a statistical improbability. Certainly not random distribution. On Jan 6, 2013, at 4:36 PM, Nick Khamis sym...@gmail.com wrote: Hello Osama, and Hisham, At 1330GMT there was some malicious activity coming from your network IP 37.75.210.90. Please act accordingly. Things that may be of use 972599779558 N. -- _ -- 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] Paltel subscribers as called parties for SIP attacks
On Aug 6, 2013, at 2:59 PM, Chris Bagnall aster...@lists.minotaur.cc wrote: FWIW, we routinely see dodgy traffic from: ovh.net hetzner.de But since those are 2 of the larger short-term contract dedicated server vendors, I'm not surprised about that. It's so frequent that I don't even bother reporting it any more - when an abuse report is acted upon and the server shut down, another pops up to take its place. all going to 972-59-* numbers (i.e. Paltel/Jawal mobile customers). Likewise here. Well, not all, but a sizeable percentage of it. We're based in the UK. Why would an internet subscriber from hadara.ps, for instance, want to call a Paltel mobile user via some remotely hacked SIP PBX thousands of miles away given than Paltel is partially owned by Hadara Technology Investment Co. (and Paltel leases long-haul infrastructure from Hadara anyway)? Are you perhaps reading too much into it? There are insecure servers and computers all over the internet. These are (ab)used and co-opted into botnets which are in turn used to compromise SIP servers. I suspect that it's probably a financial goal (free calls, or substantial termination payouts) rather than a political goal the perpetrators are seeking. Assuming that were true, then the financial goal would be uniformly distributed since other countries would have subscribers motivated by the same set of conditions. But the high concentration of requests going to a specific region mean that there's another factor at play. And it's axiomatic in intelligence that there are no coincidences. ;-) I'd be curious to know what everyone else's experiences have been like, and why 95% or better of the SIP attacks on my PBX are destined for Paltel mobile subscribers. Perhaps the termination payout on those numbers is particularly good, and/or regulation/investigation into abuse isn't so good? Kind regards, Chris Ok, let's say it's higher than any other country. Then what? Once the art of hacking PBX's for free calls is perfected, shouldn't it trickle down into other markets where the reward is less, but someone else has already done the hard part for you? That 4 years later the overwhelming majority of calls continue to be destined to Paltel indicates that there are motivators unique to this region. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peer SIP authentication with Taqua switch
I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?) switch. I'm using a config that looks like: [sip_proxy-out] type=peer authuser=208 remotesecret=xyzzy qualify=100 host=n.n.n.n call-limit=5 nat=no ; sendrpid=yes insecure=no But the Taqua responds to outbound INVITES with 403 Forbidden (oddly, not 401 or 407). Also, from what I can tell, the outbound INVITE doesn't seem to have any fields that would imply authentication (unless console SIP debugging strips sensitive fields from output). What am I missing? Is there a good configuration and/or troubleshooting guide? I looked on voip-info, etc. but most of that covers client/server configurations, not trunk peering. Anyone have a config that they managed to get working? Thanks, -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting to a Taqua switch
Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN via SIP on a Taqua 7000 switch? My local carrier recently upgraded software and changed their configs so that signalling and media are on different cards (and hence different IP addresses), and it's causing issues. I suspect there are other factors at play... it may or may not be behind a properly configured SBC. Thanks, -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Whither app_nv_faxdetect
Anyone know where the sources for app_nv_faxdetect officially live? I couldn't turn them up on a web search, just patched versions for 1.4, etc. 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
Re: [asterisk-users] SIP TOS Not being set
On 7/25/10 7:54 PM, Nick Brown wrote: Hi All, Facing an issue at the moment with setting the TOS on packets - the documentation is a bit light, however is straightforward so unsure if this is a configuration issue or a bug. Following is set in sip.conf; tos_sip=CS3 tos_audio=EF And is reflected in the CLI; IP ToS SIP: CS3 IP ToS RTP audio: EF However a packet capture shows the following; RTP Packet looks good; 11:39:59.554679 IP (tos 0xb8, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) LOCAL.26392 REMOTE.8768: UDP, length 172 Signaling Packet not so good; 11:39:59.633869 IP (tos 0x0, ttl 64, id 35957, offset 0, flags [none], proto: UDP (17), length: 479) LOCAL.sip REMOTE.sip: SIP, length: 451 Seeing the same behavior on 1.4.28 and 1.6.2.9, separate servers. The packet captures are from the box itself so will not be affected by anything upstream. Anyone able to advise if they see the same problem? Cheers Nick. Seeing it on trunk, as well: 20:39:46.084309 IP (tos 0x0, ttl 64, id 0, offset 0, flags [DF], proto: UDP (17), length: 200) 66.232.79.143.14572 66.232.80.9.49152: [udp sum ok] UDP, length 172 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
On 7/24/10 1:23 PM, Ira wrote: At 10:01 AM 7/24/2010, you wrote: The version of gcc is most definitely related to this problem, although it is the linker that is the problem. My recommendation for solving this problem is to use 'make menuselect' to eliminate any modules that you do not wish to load, then ensure modules that you definitely want (app_stack, app_voicemail) are selected. Follow that up by eliminating all noload statements from /etc/asterisk/modules.conf and Asterisk should load fine. I wonder if this is a problem with my old modules.conf. I'll rename it and see if that clears up the problem. Ira This might not be the simplest solution, but it's the one that I've used and it's been very reliable. I usually do the following just before editing a config file for the first time: % cp -p /etc/asterisk/foo.conf /etc/asterisk/foo.conf.orig Or conversely, right after an install but before any customization: % mkdir -p $HOME/asterisk % cp -a /etc/asterisk $HOME/asterisk/conf-1.6 Prior to a version bump, rename your config directory, i.e. % /etc/init.d/asterisk stop % mv /etc/asterisk /etc/asterisk-1.6 Diff your modified config files against the pristine versions. If you backed up the originals, then you can diff them one-off against their .orig versions... otherwise, you can do a recursive side-by-side diff of your running system and a pristine backup you made: % diff -ur $HOME/asterisk/conf-1.6 /etc/asterisk-1.6 ~/asterisk-config.diff Now install your new version of asterisk (1.8 or trunk or whatever), make a pristine copy of /etc/asterisk as above, and hand apply your patches from the above file into /etc/asterisk. Start up asterisk manually: % /etc/init.d/asterisk start and watch for errors. To be sure, do: % grep asterisk /var/log/messages With a config of moderate complexity (not a lot of peers, but some dialplan trickiness) this takes me about 30-40 minutes. Good luck. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POE Splitters
Sounds like a great ear warmer!!! Hell, you can probably grill a panini with it if you're patient. On 7/23/10 6:39 AM, Matt wrote: You're using phones that draw 15Watts?!?! Let me know what brand this is so I can stay away from them. On Thu, Jul 22, 2010 at 2:58 PM, David Gibbons d...@videon-central.com mailto:d...@videon-central.com wrote: There is no such device -- it's outside of the POE spec. Class 3 devices are allowed to consume at max 15.4W. Most phones are class 3 devices. The math just doesn't work out. Even if you used the draft standard for class 4 (~30W), you could still power max 2 devices at 15W/ea. -Dave On Thu, Jul 22, 2010 at 2:46 PM, Matt mhop...@gmail.com mailto:mhop...@gmail.com wrote: I've got an interesting situation where I have one cable run from the feed area to the service area. I have three devices that I need to power at the service area. Is anyone aware of a device that will take the POE from the cable run and then allow me to split it to two or three devices at the service end? When I search for splitter all I get are the injectors, but I figure someone has to make something I realize I'll need a power adapter with enough amps to power the full load at the end. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.0-beta1 is Now Available!
On 7/23/10 6:18 PM, Ira wrote: At 02:58 PM 7/23/2010, you wrote: The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. So being the brave type, I downloaded and installed this onto my Asterisk Box. Compiled fine and installed fine, but it didn't work. I kept getting errors on gosub and none of my DAHDI channels were visible. So I went back to 1.6.2.11-beta one and all was well again. Is there something really basic I missed to get 1.8 to work? Ira What sort of errors on your Gosub's? -- _ -- 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] US Truth in caller id act... and it's impact onservices
Well, I don't want to take this too much off-topic, but intent is in law an extremely difficult charge to prove. You basically need to have witnesses confirm that the accused told them unambiguously that this is what he had intended to do. It is one of the most challenging prosecutions to make, unless you basically had the accused come out and admit it. Toll carriers basically trust the caller-id they are given, even though that call is originating outside of their network. Hence the precedent exists, in that the argument could be made that the local access leg to the toll-carrier is one call, and the toll transit leg is another call... they just happen to be connected end-to-end. On 5/25/10 8:23 PM, j...@j4computers.com wrote: I submit this is still very ambiguous. The intent to . . . deceive is, like beauty all in the eye of the beholder, or, (or accuser) one must be able to accurately read the mind of the perpetrator. Also, what is inaccurate? Is that when you transmit the original ID, as is default? Is that accurate? Or is it accurate to say it came from the hairpin machine? A field day for lawyers. As are so many laws. Wiggle room. joe a. i On 5/22/2010 at 3:28 PM, Philip Prindeville philipp_s...@redfish-solutions.com wrote: For the 3rd consecutive term, the US Senate has introduced the Truth in caller ID Act of 2009. It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out or hairpin (without a redirect) to the PSTN (therefore as two separate but bridged call legs) and put the caller ID of the 1st call onto the 2nd leg (which is, by the way, the default behavior of Asterisk) you may be breaking the law in some states. This law introduces uniformity across all states (it's nice to have a level playing field, whether you agree with this law or not). It also very specifically defines under what condition spoofing/swatting is illegal: (1)IN GENERAL- It shall be unlawful for any person within the United States,in connection with any real time voice communications service, regardless of the technology or network utilized, to cause anycaller ID service to transmit misleading or inaccuratecaller ID information, with the intent to defraud or deceive. http://thomas.loc.gov/home/gpoxmlc111/h1258_eh.xml which is nice, because it's less ambiguous about when the activity is illegal (and avoids unnecessary contention between customers, telcos, and PUC's). For instance, if you're implementing single number calling for your employees, so that their cell-originated calls indicates their primary (deskphone) work number, the the intent to defraud or deceive is absent. This act delivers a badly needed brightline definition of what can and can't be done within the limits of the law. If you agree with this law, and believe that it facilitates the deployment of useful calling features, then please contact your congressman. And if you don't, well, you have a voice too, so tell them why it falls short. Either way, this act has been backburnered way too long and it's time to have a final conclusion on the matter. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] US Truth in caller id act... and it's impact on services
For the 3rd consecutive term, the US Senate has introduced the Truth in caller ID Act of 2009. It was passed by the Senate (finally) in January, and has moved to the House for a vote. A lot of states have ambiguous or overly restrictive language on how caller ID may be manipulated. For instance, if you have a PBX, and a call comes in from the PSTN, which you then loop back out or hairpin (without a redirect) to the PSTN (therefore as two separate but bridged call legs) and put the caller ID of the 1st call onto the 2nd leg (which is, by the way, the default behavior of Asterisk) you may be breaking the law in some states. This law introduces uniformity across all states (it's nice to have a level playing field, whether you agree with this law or not). It also very specifically defines under what condition spoofing/swatting is illegal: (1)IN GENERAL- It shall be unlawful for any person within the United States,in connection with any real time voice communications service, regardless of the technology or network utilized, to cause anycaller ID service to transmit misleading or inaccuratecaller ID information, with the intent to defraud or deceive. http://thomas.loc.gov/home/gpoxmlc111/h1258_eh.xml which is nice, because it's less ambiguous about when the activity is illegal (and avoids unnecessary contention between customers, telcos, and PUC's). For instance, if you're implementing single number calling for your employees, so that their cell-originated calls indicates their primary (deskphone) work number, the the intent to defraud or deceive is absent. This act delivers a badly needed brightline definition of what can and can't be done within the limits of the law. If you agree with this law, and believe that it facilitates the deployment of useful calling features, then please contact your congressman. And if you don't, well, you have a voice too, so tell them why it falls short. Either way, this act has been backburnered way too long and it's time to have a final conclusion on the matter. -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peering with a Taqua T7000
Anyone have any luck configuring a SIP trunk on a Taqua to talk to Asterisk? We were initially set up as a subscriber (access line) but that had some undesirable side-effects, such as quashing the ANI on outbound calls. Looks like we're going to have to reconfigure the trunk as a network gateway. I asked their Director of Product Management for product documentation but didn't hear back, so I guess we're on our own. If anyone else has successfully interoperated, please share your results. Also any information about Diversion: or P-Asserted-Identity: results would also be handy. Thanks, -Philip -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issue with (pattern) matching extension
On 4/29/10 4:22 AM, Jim Dickenson wrote:
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
That's capital N, isn't it?
Also, the prefix _stdexten-. seems to work fine in the [stdexten]
context, so I'm not sure what's different here.
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Re: [asterisk-users] Issue with (pattern) matching extension
On 4/29/10 1:55 PM, Tilghman Lesher wrote: On Thursday 29 April 2010 13:11:17 Philip A. Prindeville wrote: Doesn't quite make it 'deterministic' if you have to test it to see what it's going to do. The code is deterministic. The human who wrote the example is not. Are you proposing a genetic modification to make humans deterministic? If we're going to examine gene therapy, let's start with suppressing the polemic gene, shall we? Rather than your committing a point fix to stdexten which was reported as a side-effect, you might also have looked over my proposed fix which covered both, reviewed that, and committed that instead. Oh, well. -- _ -- 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] Some progress, anyway...
Just saw from build 2036: Starting mini_httpd... WARNING WARNING WARNING YOU STILL HAVE NOT CHANGED YOUR HTTPS ADMIN PASSWORD ANYONE THAT KNOWS YOU ARE USING ASTLINUX CAN DESTROY YOUR SYSTEM. PLEASE CHANGE THIS OR DISABLE THE HTTPS ADMIN INTERFACE IMMEDIATELY! Example: htpasswd /var/www/admin/.htpasswd admin WARNING WARNING WARNING Starting mini_httpd (HTTP only)... cat: can't open '/tmp/mydhcpip': No such file or directory This is pbx (Linux i586 2.6.25.19-astlinux) 11:30:27 pbx login: Now, to get the following packages to build: misdn asterisk-chanmisdn nistnet rhino strace rp-pppoe ___ -- 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] Some progress, anyway...
Darrick Hartman wrote: Philip Prindeville wrote: Just saw from build 2036: Now, to get the following packages to build: misdn asterisk-chanmisdn nistnet rhino strace rp-pppoe Whoops. I'm sure Philip thought he was sending this to a different mailing list. Indeed... another infamous (well, not quite) autocompletion blooper. Never mind. -Philip ___ -- 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] Troubleshooting one-way voice... how to peek into SIP RTP?
Andres wrote: I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Use tcpdump to capture to a file both call scenarios. Then use Wireshark to open the file. You can then do an 'RTP- Show All Streams' Analysis of the calls. That alone would reveal whether the Audio is really there or not. You can export that G711 Payload and listen to it with the Windows Media Player. I'm running wireshark 1.0.3. I've opened the captures... How do I examine the streams? I don't follow what you're saying above. And does anyone have a plugin that would allow actual playback of the .pcap files' audio packets? Thanks, -Philip If you don't see the RTP in one direction then you might have a signalling problem. Andres http://www.neuroredes.com Thanks, -Philip ___ -- 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
Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
Philip Prindeville wrote:
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
exten = 208345,n,Goto(redfish-pstn,s,1)
...
[redfish-pstn]
exten = s,1(incoming),Noop()
exten = s,n,Answer()
exten = s,n,Wait(0.5)
...
some filters for bogus ANI's like 8 goes to badani below
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,nWaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
exten = s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing)
exten = s,n,Playback(privacy-unident)
exten = s,n,Wait(0.5)
exten = s,n,Congestion()
exten = s,n,Hangup()
include = redfish-extens
exten = i,1,NoOp(Invalid: ${EXTEN})
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[redfish-extens]
...
exten = 113,1,Monitor(wav,,w); for debugging
exten = 113,n,Macro(stdexten,113,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
...
exten = 113,1,Macro(stdexten,119,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
Err, sorry. Typo. That was:
exten = 119,1,Macro(stdexten,119,${GUEST},redfish)
exten = 119,n,Goto(s,exten)
-Philip
So I don't get this at all. If I dial 208345, then enter '119' as
the extension, it rings on a few phones (including a Xlite softphone)
and if I pick up on any of those, I get one-way voice (I can hear the
caller but they can't hear me).
If I enter '113' as the extension, it rings on two SPA-942's (one of
which is the same as above, just a different line presentation)... and
if I answer, then I get two-way voice! Only difference is the Monitor()
statement.
I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why
Asterisk would need to transcode a call between two uLaw endpoints, I
don't know... and (b) why is it staying in the Media path at all?
I have the SIP peer that the calls come in on as:
[sip-proxy]
...
type=peer
nat=no
canreinvite=no
reinvite=no
Anyone know why the Monitor() would change the duplex(ity) of the audio
stream? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
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Re: [asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
exten = 208345,n,Goto(redfish-pstn,s,1)
...
[redfish-pstn]
exten = s,1(incoming),Noop()
exten = s,n,Answer()
exten = s,n,Wait(0.5)
...
some filters for bogus ANI's like 8 goes to badani below
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,nWaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
exten = s,n(badani),Log(DEBUG,ANI: ${CALLERID(ani)} clearing)
exten = s,n,Playback(privacy-unident)
exten = s,n,Wait(0.5)
exten = s,n,Congestion()
exten = s,n,Hangup()
include = redfish-extens
exten = i,1,NoOp(Invalid: ${EXTEN})
exten = i,n,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[redfish-extens]
...
exten = 113,1,Monitor(wav,,w); for debugging
exten = 113,n,Macro(stdexten,113,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
...
exten = 113,1,Macro(stdexten,119,${GUEST},redfish)
exten = 113,n,Goto(s,exten)
So I don't get this at all. If I dial 208345, then enter '119' as
the extension, it rings on a few phones (including a Xlite softphone)
and if I pick up on any of those, I get one-way voice (I can hear the
caller but they can't hear me).
If I enter '113' as the extension, it rings on two SPA-942's (one of
which is the same as above, just a different line presentation)... and
if I answer, then I get two-way voice! Only difference is the Monitor()
statement.
I'm starting to suspect it's a CODEC issue in Asterisk, though (a) why
Asterisk would need to transcode a call between two uLaw endpoints, I
don't know... and (b) why is it staying in the Media path at all?
I have the SIP peer that the calls come in on as:
[sip-proxy]
...
type=peer
nat=no
canreinvite=no
reinvite=no
Anyone know why the Monitor() would change the duplex(ity) of the audio
stream? I'm baffled (no pun intended). And is there any debugging I
can turn on to reveal CODEC behavior that might differ from 113 and 119?
Thanks,
-Philip
Philip Prindeville wrote:
I've got the following situation. I'm running Asterisk 1.4.18 on a
firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones
behind it.
I'm peering SIP with a Coppercom switch sitting behind an SBC.
On outbound calls, I get 2-way voice, no worries.
On inbound calls, I get one-way voice (I can hear the caller but they
can't hear me).
I've looked at tcpdumps of the RTP traffic, and the addresses and port
numbers correspond to what's in the SIP INVITE/OK messages (assuming
that they don't somehow get munged by NAT after tcpdump looks at them --
there is no NAT device upstream of my Asterisk firewall).
I'll look into using Record() or Monitor() to capture the phone call,
but if there's any conversion being done by codecs then that won't
eliminate the possibility that the code itself is misconfigured or buggy
and generating a bad stream on one of the legs...
Anyone have an idea about how to best go about troubleshooting this?
Thanks,
-Philip
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[asterisk-users] Troubleshooting one-way voice... how to peek into SIP RTP?
I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip ___ -- 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] What replaces Macro() now? And how do you do the equivalent?
I've been working on getting the sample configuration of extensions.conf to be more usable, i.e. to make the examples be more flexible or cover more territory... I thought it might be handy to show people how to use more contexts for virtual hosting, for example. Problem is I was using the existing stdexten macro from 1.2. See: http://bugs.digium.com/view.php?id=11969 If Macro()/MacroExit() is deprecated, how does one go about achieving the same functionality with Gosub()/Return()? Not having named parameters is a bit of a hassle. I was thinking about setting up variables, and using a template, but that's clumsy also. So... if we're giving up Macro(), what functionality do we get instead? Thanks, -Philip ___ -- 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] One server, multiple companies
Sorry for the late follow-up to this... it was on my to-do list for
over a month... Sigh.
I've submitted a configuration bug and for this:
http://bugs.digium.com/view.php?id=11969
The hope being that if the examples provided in the configs/ directory
work better out of the box for real-world applications (like virtual
hosting), then more people will benefit... thus bringing world peace and
ending global warming.
If you want to encourage the bug marshals to approve these changes and
get them into the source tree, I won't discourage you.
Oh, and for the record: in my sip context for peering with my PSTN
carrier, I do a Goto the appropriate context (acme-incoming,s,1 etc)
based on the DID / DNIS coming in from the carrier... something akin to
what Jerry is doing below, but slightly more complicated (some companies
have a single shared outside number, others have individual DID's per
extension, etc).
-Philip
Jerry Jones wrote:
[incoming]
exten = 2125551211,1,GoTo(companyA,1)
exten = 2125551212,1,GoTo(companyB,1)
exten = 2125551213,1,GoTo(companyC,1)
[companyA]
exten = 2000,1,Dial()
[companyB]
exten = 2000,1,Dial()
[companyC]
exten = 2000,1,Dial()
On Dec 13, 2007, at 5:53 PM, Diego Andrés Asenjo González wrote:
-- Mensaje reenviado --
From: Eric C. [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sun, 9 Dec 2007 19:55:51 -0500
Subject: [asterisk-users] One server, multiple companies
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three
distinct phone systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan
within the extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then
using a gotoif to get to correct greeting (correct company).
My question is... lets assume all three companies have extension
numbers being 2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I
use context correctly so that the user at 4002 cannot get through
to the user at company A whose extension is 2000 as currently, I
can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my
architecture correct? Should I be running different processes for
each company? Can context resolve what I need?
Hi,
You should try DeStar, a management interface for Asterisk:
http://destar.berlios.de/
DeStar supports Virtual PBXs, then you can install it and take a
look at the dialplan. Sorry for the late answer but I've just read
the list messages.
Bye,
Diego Andrés.
So
Please advise.
thanks,
Otto
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[asterisk-users] OT: 3rd party SMS service?
I'm currently getting SIP trunking from my PSTN provider, but they don't quite grok the whole any-service/any-device philosophy... I'm wondering if it's possible to get SIP voice carriage from one provider, but have SMS associated with the same phone numbers being provided by another carrier? Or does SMS and the single source of truth that the GTT provides forbid delegation of services? -Philip ___ -- 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] OT: Recommendation for EDGE/3GPP SIP phone?
I'm looking to ditch my GSM phone with Cingular and provide my own calling services with Asterisk hanging my phone off the Edge or 3GPP network... Luckily, I have the option (through my employer) of getting a data-only plan. My question is, other than the Nokia E61i or E70, what phones will do SIP-over-EDGE or SIP-over-3GPP? The iPhone doesn't yet have a SIP client, and it's EDGE-only. There are probably knowledgeable folks out there that have tested/used such devices. Bonus points for one that's WIFI capable (I don't think my data plan is all you can eat, and Cingular coverage is far from complete). Thanks, -Philip ___ -- 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] Multiple SIP phones behind a Linksys firewall
Luki wrote:
I always assumed that you can have multiple SIP phones behind a Linksys
firewall/router (WRT54G) all using the same STUN server/port.
I got 10-20 SPA942's behind a OpenWRT router (on WRT54G, WRTSL54GS,
...) at several sites, no STUN, no special configuration, no problems
at all. Just as a precaution, I set the SIP port and RTP port range
for each phone differently so that it's unique (i.e. Phone 1 SIP port
6001 and RTP 10100-10199, etc.) but that's really just a precaution to
help the the Linux' conntrack on the OpenWRT a bit. It's not really
needed as the router will resolve port conflicts by rewriting the
ports transparently.
Bottom line, a few phones behind a well-behaved NAT should work just fine.
/Luki
What do the iptables look like on OpenWRT? Are they configured as part
of the release, or left to the user to configure, or what?
I'm using a Soekris net5501 running Astlinux 0.5 trunk (with a patched
version of Arno's firewall script that has not yet been integrated into
the source tree): it supports the ip_conntrack_sip and ip_nat_sip modules.
I have the firewall/Asterisk running on this box at the home office,
with a couple of SPA's behind it (942's and a PAP2-NA).
Then I have remote offices also with SPA-942's sitting behind a
similarly configured Soekris 942 (only difference being that Asterisk
isn't running on it).
I had all of the usual NAT related issues (one-way audio, no audio, etc)
until I patched in the NAT SIP modules.
I've attached it. This works with arno-iptables-firewall-1.8.8l.
Arno says he's working on a plug-in for 1.8.8m and 1.9.0? that will be
released separately, but I've haven't yet seen it.
-Philip
--- ./arno-iptables-firewall.sipnat 2008-01-22 01:10:19.0 -0800
+++ ./arno-iptables-firewall1980-05-02 00:31:28.0 -0700
@@ -348,6 +353,14 @@
# write rules matching the state of a
connection
module_probe ip_conntrack_ftp# Permits active FTP; requires
ip_conntrack
+ if [ -n $SIP_PORTS ]; then
+ports=
+for port in $SIP_PORTS; do
+ $ports=$ports${ports:+,}$port
+done
+module_probe ip_conntrack_sip ports=$ports
+ fi
+
module_probe ipt_conntrack # Allows tracking for various
protocols, placing entries
# in the conntrack table etc.
module_probe ipt_limit # Allows log limits
@@ -393,6 +403,10 @@
if [ $NAT = 1 ]; then
#module_probe iptable_nat# Implements nat table
module_probe ip_nat_ftp # Permits active FTP via nat; requires
ip_conntrack, iptables_nat
+if [ -n $SIP_PORTS ]; then
+ module_probe ip_nat_sip
+fi
+
module_probe ipt_MASQUERADE # Implements the MASQUERADE target
fi
@@ -3191,9 +3205,9 @@
# Adding UDP ports NOT to be firewalled
###
- if [ -n $OPEN_UDP ]; then
+ if [ -n $OPEN_UDP -o -n $SIP_PORTS ]; then
echo Allowing the whole world to connect to UDP port(s): $OPEN_UDP
-for port in $OPEN_UDP; do
+for port in $OPEN_UDP $SIP_PORTS; do
$IPTABLES -A EXT_INPUT_CHAIN -p udp --dport $port -j ACCEPT
done
fi
--- ./etc/arno-iptables-firewall/firewall.conf 2007-12-17 10:30:55.0
-0800
+++ ./etc/arno-iptables-firewall/firewall.conf.new 2008-01-28
09:47:37.0 -0800
@@ -1134,3 +1134,7 @@
# should always contain a carriage-return (enter)!
# -
#BLOCK_HOSTS_FILE=/etc/arno-iptables-firewall/blocked-hosts
+
+# Specify UDP ports used by Asterisk registration end-points or by SIP
+# phones (8 max).
+#SIP_PORTS=5060 5061 5062 5063 5064
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[asterisk-users] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
Howdy, Excuse the neophyte questions... I was wondering: (1) what's involved in setting up a call with encrypted media (I'm on a cable network and don't want my calls snooped); (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? (3) my PSTN service provider that I have SIP trunking to doesn't provide SMS service (yet or possibly ever)--is there a way to shop-out SMS for my associated numbers from someone else? I eventually hope to have a cell phone with dataplan only that I then do SIP-over-UMTS (like a Nokia E70 or E61i, for instance)... if I'm moving towards having a single number, then I figure text messaging should work regardless of the device I'm using (softphone on a laptop, VoWifi handset, or POE hardphone). Any service/any device, right? That's what it's all about. (4) I have an HP all-in-one officejet and a SPA-2000 (or is it an unlocked PAP2-NA?) that I can use for fax service, but was wondering if I could use my Asterisk box (it's a 400MHz Geode LX) as a fax server without too much impact. Thanks, -Philip ___ -- 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] Can't read environment variable
Uhhh... just
export HOSTNAME
should be enough once it's been set.
Joost Kuif | Mobillion wrote:
This pointed me into the right direction, thanks Tzafrir!
i added a export HOSTNAME=$HOSTNAME into my .bash_profile
Grtz,
Joost
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Tzafrir Cohen
Verzonden: Wednesday, January 30, 2008 1:58 PM
Aan: asterisk-users@lists.digium.com
Onderwerp: Re: [asterisk-users] Can't read environment variable
On Wed, Jan 30, 2008 at 01:38:48PM +0100, Joost Kuif | Mobillion wrote:
Hi,
I can't read a environment variable in a asterisk dialplan.
When logged in as user root on the system an 'echo $HOSTNAME' gives
the hostame of the machine.
Asterisk (1.4) is started from the same console.
I try to read it like this:
exten = s,n,NoOp(host=${ENV(HOSTNAME)})
Does anyone know what i am missing?
Is that variable set?
cat /proc/PID_OF_ASTERISK/environ | tr '\0' '\n' | grep ^HOSTNAME=
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Re: [asterisk-users] Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)
Chris Bagnall wrote: (2) is there a cheat-sheet for configuring Sipura handsets/hardphones like the SPA-942, and in particular for message-waiting indicator and shared-line appearances? MWI is easy... simply add a [EMAIL PROTECTED] setting to sip.conf for the phone Make sure also to add voice mail number under admin/advanced/Phone menu on the 942. It does not seem to pick it up automatically from asterisk like many phones do. If you’re configuring a lot of similar handsets, consider using an autoprovisioning script - it'll save you a hell of a lot of time in the long run. Regards, Chris I've thought about doing that... My main two issues are: (1) it's a pain not being able to stuff an XML file via http into a Sipura (I think you either have to use TFTP or else HTTPS), and (2) not having a single source of configuration state to put into the phones. I could grab state out of multiple places, but besides being messy, that also leads to things getting into disagreement when they are edited in one place but not others, etc. I've also not found a good cheat sheet that says what fields need to be set to what. I can do the scripting/programming no problem. What I don't have time for is the learning curve of figuring out from scratch what settings work best. I figure someone else out there has already done that quite effectively. -Philip ___ -- 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] When does Asterisk REFER?
I was wondering under what conditions Asterisk will hand off a call to another switch. I'm trying to verify that my local PSTN's Coppercom switch operates correctly... and wanted to know how to get a call REFER'd to another end-point. Thanks, -Philip ___ -- 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] Newbie Q: Good link to configuring NAT with Sipura ATA's hardphones
In trying to get my services up and running, I've encountered the usual spate of first-time issues. I was wondering if there was a good FAQ or Howto on troubleshooting NAT issues. The equipment that I'm using is typically a Sipura ATA or hardphone (SPA-942) sitting behind either a Linux box as firewall (using IPtables and Arno's firewall) or else a Cisco IOS device running 12.4T and Advanced Security featureset (w/ inspect configured). The SIP proxy is an Asterisk box with a public/routable interface running 1.2 or 1.4 on Linux. Usual issues include: * the first line on port 5060 doesn't register, etc. but devices on 5061-5080 work fine (caused by IOS interference) * or else SIP signalling works fine, but there's no media connection. I've googled various postings on the net, but they're not very consistent or even well explained. Some claim using VIA works, others espouse using rport=, etc. I've looked at the Sipura administration guide, and it's troubleshooting section covers 3 scenarios (!!!): * phone won't boot up * phone won't make or receive calls * calls with poor voice quality Ok, well, that leaves a lot of territory uncovered... I'd be happy to write up a FAQ once someone has explained things to me. Thanks, -Philip ___ --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] [Zaptel] Why no port to Windos?
Lee Jenkins wrote: Vincent wrote: On Fri, 14 Dec 2007 10:51:10 -0500, Lee Jenkins [EMAIL PROTECTED] wrote: I have to reboot my desktop xp box daily for it to run well. I haven't rebooted my XPSP2 in months, and I let it run 24/7, with a bunch of apps open at all times. And this is a 300E no-name box. If your PC is so unstable, you should investigate the hardware and/or the device drivers. Maybe. Its not that its unstable, the system just becomes progressively slower and less responsive if I don't reboot once in a while. I also run scandisk and defrag weekly. Of course, it may have just as much do with the type of apps that I have open and running all the time as well. As I said, I like Windows, but I don't see a Server 2000 box out performing a comparable linux box for larger pbx systems. A small office, sure. I wonder if the linux box was also running Gnome or some other desktop at the same time, would that make it a closer comparison? Maybe Windows would outperform the linux box then? Part of the difference in stability in Linux vs. Windows from what I can tell has to do with the extensive use of threads in Windows. Threads basically live for ever, and in a shared address space/container. Processes also mean that there's an upper bound on how long any sort of memory leaks can persist. Versus just spawning a process, having it work, then exit (and free up all resources with no leaks and no residual fragmentation of the heap) Here's a suggestion: try getting into your registry, find the services that seem to be resource hogs, and try splitting them out into their own instances of svchost.exe. For the non-essential services (which are most), you can restart them periodically and that will clean things up a bit. I'm not an expert, but there are resources out there on the web about how to repackage a server for increased stability. Gnome versus the Windows desktop isn't a useful comparison either. The desktop is run cooperatively by all processes, and unstable process can pretty much trash the internal state of the desktop for everyone. Not so with X Windows. You can be greedy and use up all of the resources (backing store, graphics contexts, etc) but since most useful stuff is associated with a window or group of windows, and windows are owned by a process... if that process exists, its windows (and their associated resources) usually get cleaned up. Again, no persistent damage done by a process gone amuck. Very different from the threaded/shared memory architecture of Windows. It's potentially much more efficient (emphasis on potentially)... but it's also a lot more vulnerable to misbehaving applications. -Philip ___ --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] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
Philipp Kempgen wrote: Anselm Martin Hoffmeister wrote: In most cases it seems to end at the fact that providers correct caller-ids they get from the calling party: If you send any number which is assigned to the PRI (or SIP trunk), that is fine; if you send another number, it will be changed to the (first) number of the PRI/trunk. Few providers allow for foreign caller ids to be sent over their equipment - in some countries this is even illegal. For example, one of my providers (German) allows to set any CALLERID, but their documentation warns to not do stupid tricks, as calls can be tracked and using malicious information will be prosecuted. This feature is to be used only for sending _my_ cell phone number etc. Do you know of any GSM providers/contracts where faking for a valid reason is possible? Regards, Philipp Kempgen I can think of some... in rural Idaho, cell coverage is sparse. I might check my voice mail of my cell phone via a land line, and want to call back with a response originating from my cell number... Also the case if I have my cell set to forward-on-busy to my land line, or if I'm hosting an answering service, etc. -Philip ___ --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] Newbie question: how to proxy the *real* caller-id on find-me/follow-me
I've got the following set up:
Someone calls into my PBX on a single number (via SIP trunk from my
carrier), and the get a voice menu of extensions.
On one of the extensions, it rings a bunch of internal SIP hardphones,
plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN
gateway.
The issue is that my cellphone shows my PBX's number, not the original
calling number.
My dialplan looks like:
[globals]
...
TRUNK=SIP/sip_proxy-out
CELL=${TRUNK}/208xxx
PHILIP=SIP/bedroom_1SIP/office_2SIP/kitchen_1${CELL}
[incoming]
exten = s,1,Answer()
; sometimes signaling and media get out of sync on cell networks...
exten = s,n,Wait(0.75)
exten = s,n,Playback(main-menu)
exten = s,n(exten),Background(vm-enter-number-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup
...
exten = 111,1,Macro(stdexten,111,${PHILIP})
exten = 111,n,Goto(s,exten)
exten = 112,1,Macro(stdexten,112,${REDFISH})
exten = 112,n,Goto(s,exten)
...
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
Ok, so far, so good.
The problem is that when we hit Macro(stdexten,111,${PHILIP}) and it
does the Dial(${PHILIP}) which includes the
SIP/sip_proxy-out/208xxx, 208xxx rings with my PBX's extension.
Oddly, the internal phones ring with outside caller's extension.
[sip_proxy-out]
type=peer
fromuser=208nnn
fromdomain=x.x.x.x
host=y.y.y.y
call-limit=5
nat=yes
So I'm not setting the callerid on the peer by default. What am I
missing? Do I need to modify the stdexten macro to dial with the 'o'
option? Or can I set this explicitly with a 'Set' before calling the
macro? Or do I need to be missing with the RDNIS?
Oh, I'm running Asterisk 1.2.25... (yes, I'll upgrade when AstLinux
upgrades).
-Philip
P.S. I tried adding |o to the end of the PHILIP variable, but this
didn't seem to make a difference.
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Re: [asterisk-users] [Zaptel] Why no port to Windos?
Doug wrote: At 19:55 12/13/2007, Vincent wrote: Hello I was wondering why there doesn't seem to a Windows version of Zaptel, making the Digium and its clones unavailable for a Windows PBX. Is the Zaptel/Zapata combo too *nix-centric? Thanks. Windows is a half-baked, dying OS that in essence is a 32 bit extension and graphical shell, for a 16 bit patch to an 8 bit operating system, originally coded for a 4 bit microprocessor, written by a 2 bit company, that can't stand 1 bit of competition. Do you really want to reboot your telephone system 3 times a day? And yet... the next time you turn on CNN and see Tomahawk missiles coming out of the vertical launch tubes of an Aegis class DDG (guided missile destroyer)... well, keep in mind that the Fire Control System is running a stripped down NT4 kernel. (It might be NT5 or 6 by now... my information is a little old on this particular subject. Then again, knowing how DoD certification works, it might not have budged at all.) ___ --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] [Zaptel] Why no port to Windos?
Tilghman Lesher wrote: On Thursday 13 December 2007 19:55:39 Vincent wrote: I was wondering why there doesn't seem to a Windows version of Zaptel, making the Digium and its clones unavailable for a Windows PBX. Because nobody has done it yet. The real answer is probably more along the lines of that there's no competant Windows device driver programmer who would be willing to expend the necessary effort to port the driver, for free. I'm sure that technically, it's possible, although certain assumptions that were made when developing zaptel may not be true when it comes to Windows. It is likely to be a very strenuous job to port the framework and all of the drivers. What drivers do you need to run in a purely SIP mode? zt_dummy for timing? What else? -Philip ___ --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] Sipura provisioning
Ok, I think I asked this previously but don't remember seeing an answer... Yes, you can tickle an SPA94x or 962 and have it fetch a config from a TFTP server... But is there no way to simply push a couple of lines of XML config to it directly via an HTTP POST (sans TFTP server)? Thanks, -Philip ___ --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 XML for configuration management, single-source-of-truth, etc.
Kristian Kielhofner wrote: On Dec 9, 2007 2:05 AM, Philip Prindeville [EMAIL PROTECTED] wrote: ..snip.. You think that an Asterisk configuration is a lot larger than a Cisco 5850 Access Server or a 7216 core router? IOS doesn't use XML for configuration. What's a 7216? Actually, it does I just don't know if that ever got exposed to the public or not. (Of course, the customers that wanted XML-based configs also wanted ION, so it might only have been exposed on ION.) While I was there (2000-2005) there was a big effort to have all config files be represented as XML. If you ever tried to diff two configs of a 7216 (VXR/CMTS) that were from different releases, you'd know why. This would drive customers crazy: portions of config would move around, whitespace would appear where it wasn't previously, names of commands would gratuitously change, etc. Parse-trees would be easier to diff. There were some interesting efforts floating around to have the configuration be sucked up via XML/secure-RPC to a client, edited there, and then pushed back into the router/IAD/firewall, what-have-you. The IOS parser was one of the hairiest pieces of code I had ever had to maintain. But we're getting off the subject. -Philip P.S. I know zip about Skinny, so don't ask... ___ --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] Dual-home Wifi/GSM phones for North America
So, for the hotel project, what was the conclusion? I don't remember seeing a summary. And were any of the phones combination Wifi/GSM? Like a Nokia E70 or E61i? I've been looking for such a phone to use myself, but so far haven't found one that I liked. Common flaws were: * poor standby time * lousy interface for doing complex configurations (esp. certificate management) * poor external provisioning documentation or requiring proprietary (and Windows-based) tools * inadequate SIP implementation * inability to hand-off between Wifi and GSM (or reverse) seamlessly Has anyone found a product that overcomes these issues? And what's the word on a Wifi/SIP client for the iPhone? -Philip ___ --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] One server, multiple companies
Eric C. wrote:
Hello all,
Just starting to setup asterisk v 1.4.11 and need to run three distinct phone
systems for three different companies.
So far, I have inbound lines going to the appropriate dial plan within the
extensions.conf file. I'm using
exten = _X.,1,NoOp(FROM NUMBER: ${SIP_HEADER(TO):5:10})
to determine which number is being dialed by the caller and then using a
gotoif to get to correct greeting (correct company).
My question is... lets assume all three companies have extension numbers
being 2000, 2001 2002, how does one separate them?
Or, lets say the extensions are:
company A -- 2000, 2001,2002
company B -- 3000, 3001, 3002
company C -- 4000, 4001, 4002
Since they're on one server with one asterisk process, how can I use context
correctly so that the user at 4002 cannot get through to the user at company
A whose extension is 2000 as currently, I can dial 2000 from phone 4002.
That's my current problem, how should this be setup? Is my architecture
correct? Should I be running different processes for each company? Can
context resolve what I need?
Please advise.
thanks,
Otto
First off, *nuke* the default context in sip.conf, extensions.conf, and
voicemail.conf ... it will just get you into trouble!
I do something like in my extensions.conf file:
[incoming]
exten = 208229,1,Goto(s,1,incoming-acme)
exten = 208229,1,Goto(s,1,incoming-fido)
exten = 208229,1,Goto(s,1,incoming-big-jims)
...
[incoming-acme]
exten = s,1,Answer()
exten = s,n,Wait(0.75)
exten = s,n(greeting),Playback(brief-directory-acme)
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
; these are the extensions that are exposed both to internal callers as
; well as to incoming calls... be careful what you put here.
include = extens-acme
exten = i,1,Playback(pbx-invalid)
exten = i,n,Goto(s,exten)
exten = t,1,Goto(s,goodbye)
[internal-acme]
exten = s,1,Answer()
exten = s,n(exten),Background(vm-enter-num-to-call)
exten = s,n,WaitExten(5)
exten = s,n(goodbye),Playback(vm-goodbye)
exten = s,n(end),Hangup()
include = outbound-toll
include = outbound-local
include = extens-acme
; for our SIP phones, we can program a non-numeric extension
exten = voicemail,1,VoicemailMain([EMAIL PROTECTED])
exten = voicemail,n,Hangup()
; and for DTMF coming through an ATA...
exten = 777,1,Goto(voicemail)
[extens-acme]
exten = 111,1,Macro(stdexten,111,${PHILIP})
exten = 111,n,Goto(s,exten)
...
[outbound-local]
exten = _NXX,1,Dial(${TRUNK}/${AREA}${EXTEN},,r)
exten = _NXX,n,Congestion()
exten = _NXX,n,Hangup()
[outbound-toll]
exten = _NX,1,Dial(${TRUNK}/${EXTEN},,r)
exten = _NX,n,Congestion()
exten = _NX,n,Hangup()
exten = _011.,1,Dial(${TRUNK}/${EXTEN:3},,r)
exten = _011.,n,Congestion()
exten = _011.,n,Hangup()
Note: we had to modify the stdexten macro to be:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
;
Re: [asterisk-users] Using XML for configuration management, single-source-of-truth, etc.
Ryan Burke wrote: Tilghman Lesher wrote: On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. That's sort of my point: that you have to reinvent it, and it's easy to get wrong. XML wouldn't make it any less wrong. There's a difference between parsing it syntactically (which XML fixes) and parsing it semantically (which XML does not). In fact, I find the configuration files, as they are now are much EASIER to parse than XML. With XML, you need to load up a whole state engine to ensure the config is properly formatted. At the simplest level, the config file as-is is simply a set of key/value pairs, which syntactically is very easy to parse. Part of the allure of the current format is also that it is human readable, which assists in manual editing. I'm not sure what part of the universe you have be from to make XML human readable (or more importantly, human-editable), but I am quite sure it is not from this planet. Well, after hand-coding HTML and SGML for 15+ years, XML isn't all that much of a stretch. More to the point though, there are some excellent schema-driven configuration managers for XML, so you wouldn't have to edit the files by hand. -Philip Can these configuration managers run from a command line? Or do they require a graphical environment? Some require X, some use curses... -Philip ___ --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 XML for configuration management, single-source-of-truth, etc.
Tilghman Lesher wrote: On Saturday 08 December 2007 00:51:44 Philip Prindeville wrote: Tilghman Lesher wrote: On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. That's sort of my point: that you have to reinvent it, and it's easy to get wrong. XML wouldn't make it any less wrong. There's a difference between parsing it syntactically (which XML fixes) and parsing it semantically (which XML does not). In fact, I find the configuration files, as they are now are much EASIER to parse than XML. With XML, you need to load up a whole state engine to ensure the config is properly formatted. At the simplest level, the config file as-is is simply a set of key/value pairs, which syntactically is very easy to parse. Part of the allure of the current format is also that it is human readable, which assists in manual editing. I'm not sure what part of the universe you have be from to make XML human readable (or more importantly, human-editable), but I am quite sure it is not from this planet. Well, after hand-coding HTML and SGML for 15+ years, XML isn't all that much of a stretch. And so can I, but to most people, XML looks like gobbleygook. BTW, 15+ years for a markup language (HTML) that has only been around for 14 years? See my postings on www-html and www-talk while it was in draft stage. More to the point though, there are some excellent schema-driven configuration managers for XML, so you wouldn't have to edit the files by hand. And that puts an additional requirement on developers, that each know how to manually (and correctly) edit the schema file to ensure that new options added are correctly specified. You haven't eliminated complexity at all, you've just shifted it to someone else. By that logic, we'd have no new operating systems, no new programming languages... Yeah, all progress entails a learning curve. Also, you're adding another tool in the chain that people must learn in order to administer an Asterisk setup. Right now, people can use whatever editor they like, whether that's vim, emacs, nano, joe, or some other editor. They learn one editor tool and can use it generically for all purposes. By adding another requirement, you're increasing the barrier to entry, which is the opposite direction that we want to go. And finally, another person has already made the point that most XML editors are graphical in nature. A great many Asterisk installations are installed in locations where a graphical front end is not practical. Not only does that mean that your options for text editors are limited, but they are unlikely to be easy to use (or learn how to use), compared to full screen text editors, which have been around for over 20 years and are fully mature software. Going back to my original posting, I was also suggesting that the parse tree from Asterisk could be read in and then dumped out as XML, so that other software could then ingest it... using it as a common format for passing configuration from one package to another. -Philip ___ --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 XML for configuration management, single-source-of-truth, etc.
Jared Smith wrote: On Sat, 2007-12-08 at 13:55 -0800, Philip Prindeville wrote: Going back to my original posting, I was also suggesting that the parse tree from Asterisk could be read in and then dumped out as XML, so that other software could then ingest it... using it as a common format for passing configuration from one package to another. If you're serious about wanting to do that, it wouldn't be hard to write a program that used the Asterisk manager interface (see the GetConfig action) to read the Asterisk configs and write them out in another format. If you were really ambitious, you could even have Asterisk get it's configs from XML by using the UpdateConfig AMI action, creating use of #exec, or by writing a realtime driver for XML. Personally, even though I'm a big fan of XML (and things like DocBook and XSLT, and having written hundreds of pages of documentation in XML), I don't see what putting the configuration in XML buys you, other than the ability to check the validity of the config file (assuming, of course, that someone writes a DTD and keeps it up to date). In a nutshell, XML is no silver bullet. -Jared Smith No, it's not (a silver bullet) -- agreed. But a lot of other devices manage their configurations via XML as well, so having a common way of representing shared state would simplify network provisioning, which was the kernel of my original posting. 3 of the handset manufacturers that I use, 1 of the firewalls, and 2 of the video-conference engines all use XML. And the list gets longer every day. Eventually, they will start to converge on common schemas as well... -Philip ___ --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 XML for configuration management, single-source-of-truth, etc.
Tzafrir Cohen wrote: [snip] 3 of the handset manufacturers that I use, 1 of the firewalls, and 2 of the video-conference engines all use XML. And the list gets longer every day. Most of the programs I have don't use XML. And I only feel better. Eventually, they will start to converge on common schemas as well... Asterisk's configuration is mostly hand-written. And also a lot larger than those of small devices on the network. You think that an Asterisk configuration is a lot larger than a Cisco 5850 Access Server or a 7216 core router? jabberd uses an XML for configuration, and I just can't make sense of it. Unnecessar-ly huge indentations, sections have to be explicitly ended, etc. Try to implement '#include' and '#exec' in a sane way with XML. You can't just include one valid XML in another. You have to make a partial XML. And apitting it out is usually way more complicated. Furthermore, there is the issue of partial processing: do you opt for one big XML file? Or continue with one XML file per .conf file? I'm fine with individual files per functionality. Makes it easier to add new functionality and keep config files forward compatible, or for that matter, to turn off or mutate individual bits of functionality by deleting or swapping out the individual configs. That said, I still wonder how to do the equivalent of apache's 'configtest' with Asterisk. Good point. -Philip ___ --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] Using XML for configuration management, single-source-of-truth, etc.
I'm starting work on some provisioning tools to simplify plugging in and configuring hard SIP handsets and conference bridges (maybe eventually MPEG-4 PoE video cameras that speak SIP as well). Issue is that I'd like to glean as much information out of the configuration files... but don't want to write a whole new parser to do it (especially not one that understands templates and macros). For instance, from the voicemail.conf, extensions.conf, and sip.conf files, I should be able to generate 90% of the configuration state needed for provisioning an out-of-the-box Sipura SPA941... if only those files were in some more parsable format, like XML. How much effort would it be to add an application that traverses the configuration state and writes it out as an XML flat file? Or perhaps at some point in the future, Asterisk's configuration files could be represented as XML natively (did someone in the back row just show gconf???). I'm a relative newbie, so if I'm missing something obvious or there's been a religious war on the subject in the past, apologies... -Philip ___ --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 XML for configuration management, single-source-of-truth, etc.
That's sort of my point: that you have to reinvent it, and it's easy to get wrong. Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. ___ --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 XML for configuration management, single-source-of-truth, etc.
Tilghman Lesher wrote: On Friday 07 December 2007 20:12:12 Philip Prindeville wrote: Darryl Dunkin wrote: You can store most of the configurations in a database which may be more accessable to you. Perl can also parse these configurations quickly enough if you know how to use the input record seperator ($/) properly. The only thing Asterisk will not store which you would probably need is the actual MAC address of the phones themselves. This may be done easily enough as comments in the users sip.conf section. That's sort of my point: that you have to reinvent it, and it's easy to get wrong. XML wouldn't make it any less wrong. There's a difference between parsing it syntactically (which XML fixes) and parsing it semantically (which XML does not). In fact, I find the configuration files, as they are now are much EASIER to parse than XML. With XML, you need to load up a whole state engine to ensure the config is properly formatted. At the simplest level, the config file as-is is simply a set of key/value pairs, which syntactically is very easy to parse. Part of the allure of the current format is also that it is human readable, which assists in manual editing. I'm not sure what part of the universe you have be from to make XML human readable (or more importantly, human-editable), but I am quite sure it is not from this planet. Well, after hand-coding HTML and SGML for 15+ years, XML isn't all that much of a stretch. More to the point though, there are some excellent schema-driven configuration managers for XML, so you wouldn't have to edit the files by hand. -Philip ___ --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] Happy Birthday Asterisk
Bill Andersen wrote: Philip Prindeville wrote: So I'd venture to say that by August, the Internet will really be *30* years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is pretty ambicious, even for Al! :) I wrote my first RFC at 22, and I was never Vice President... ;-) -Philip ___ --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] Happy Birthday Asterisk
Tilghman Lesher wrote: On Friday 07 December 2007 09:56:56 Bill Andersen wrote: Philip Prindeville wrote: So I'd venture to say that by August, the Internet will really be *30* years old. As Al Gore was born in 1948, I can see that the Internet could be as old as 30, but not much more. 35 years ago would put him at 25 years old. And inventing the whole Internet at 25 is pretty ambicious, even for Al! In actuality, most people produce all of the great inventions of their life by the time they hit 30. Einstein, for one, produced his great theory of relativity at the ripe old age of 26. Mark Spencer came up with Asterisk at age 22. So this idea that 25 is too young to produce a great achievement is baloney. BTW, Al Gore was credited with introducing the legislation that permitted commercial organizations onto the network that would become known as the Internet. So in a way, he did create the Internet, by changing the circumstances you would have to have in order to access this decentralized computer network. If you doubt the importance of having commercial organizations on the network, consider where the Internet would be, if Amazon, eBay, and Linux Support Services (d/b/a Digium) had never been allowed onto the network. Let's give credit where it's due: a lot of people in Washington were being lobbied by Bill Shrader, Vint Cerf, and Dave Van Bellengem (sp?) to be honest. All people like Senator Gore did was carry their water. The argument being put forward was that various groups (like SRI, Rand, Mitre, etc) would get onto ARPAnet because they had been awarded a DARPA or DISA or DMA contract... as would other vendors (like Boeing or Wellfleet or Raytheon...). Since SRI, Rand, Mitre, etc. would have a constant stream of contracts in progress, their ARPAnet access never went away. Others, like the latter group, would get their access yanked when their contracts ended (or got suspended while DoD budgets languished in Congress). Their argument was that such collaboration with other companies and universities would continue after contracts were completed, and that the Internet was a powerful collaboration tool (duh!)... ergo a permanent Internet was needed, even if the users had to pay for it themselves (rather than it being a perk of getting a DARPA contract). Even small companies (like FTP Software, who I was working for at the time), could benefit from being able to ship new binaries to government agencies or other partners on government contract (like HP, who we were writing a DOS TCP/IP stack for with a Sockets API... sound familiar?). In some ways, these were dark days: the future was very uncertain. On the other hand, we didn't have spam. ;-) -Philip ___ --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] Happy Birthday Asterisk
The Internet is a lot older than 20 years! I sent my first email in 1981 (yes, using SMTP and TCP/IP), and even then it had been around for a while (maybe not using TCP... but in the NCP incarnation at least). IP dates back to... what? 1978? Yeah, the RFC was published in September 1981, but implementations had been running around on IMP's, PDP-11s, and Honeywell Susie's (See IEN-106 and -111) And RFC-791 covers IP *v4*. There were earlier experimental versions of IP, but v4 got it right. So I'd venture to say that by August, the Internet will really be *30* years old. -Philip Steve Totaro wrote: Happy Birthday Internet! 20 years and there was a nice celebration with the founding fathers (visionaries) in DC. Thanks, Steve Totaro Dean Collins wrote: Sorry if this dupes, didn’t seem to post to the list when I emailed it a few hours ago. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). *From:* Dean Collins *Sent:* Thursday, December 06, 2007 2:49 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* “Happy Birthday Asterisk” So I got to drop in on the afternoon sessions of the http://www.voicepeeringforum.com http://www.voicepeeringforum.com/ held in New York yesterday. There’s a blogpost here http://deancollinsblog.blogspot.com/2007/12/vpfan-interesting-yet-frustrating.html but really just wanted to say because I didn’t see anyone else say it yesterday *“Happy Birthday Asterisk”.* Kevin Fleming from Digium gave a session yesterday about Asterisk when we all realized that Asterisk was 8 years old as of purely by co-incidence yesterday December 5th. On December 5th 1999 Mark Spenser released version 0.1.0 . and an industry was born :) I don’t know if the guys at Digium celebrated in anyway but maybe there can be a big “User Event” in Huntsville in 2 years time on the 10^th anniversary. Just a thought. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ___ --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 ___ --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 ___ --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] New feature: calling all bug marshals
So, am I not summoning the bug marshals correctly? Or like druids and
wizards, do they mystically sense when they are needed and emerge from
the deep woods just in the nick of time?
(Ever notice that whenever Batman is needed at night, it's always
cloudly? How lucky is that! What if you need Batman, and it's a
perfectly cloudless night? Or what if it were foggy?)
Will they send me a sign if my feature gets approved? Should I look out
the window towards downtown? ;-)
Or... am I really supposed to file a bug after all?
Philip Prindeville wrote:
Hi.
I wanted to write a popcorn app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten = s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten = s,n,While(${EPOCH} ${FUTURETIME})
exten = s,n,Wait(0.01)
exten = s,n,EndWhile()
exten = s,n,Play(beep)
exten = s,n,Hangup()
and hating myself for it (my Asterisk runs on a 500MHz Geode LX).
So I decided it would be useful (in general, and educational for me in
particular) to write a WaitUntil() application instead.
Well, I've done that.
I was going to file a bug and then post a fix to get their feature
in, but the Bug guidelines seem to be pretty clear that this is not
the way to go.
So, I'm posting here instead.
The example to paste into the documentation or extensions.conf is:
[popcorn]
exten = s,1,Answer()
; the amount of delay is set for English; you may need to adjust this
time
; for other languages is there's no pause before the synchronizing beep.
exten = s,n,Set(FUTURETIME=$[${EPOCH} + 11])
exten = s,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten = s,n,SayPhonetic(z)
exten = s,n,SayUnixTime(${FUTURETIME},,HNS)
exten = s,n,Playback(local)
exten = s,n,WaitUntil(${FUTURETIME})
exten = s,n,Playback(beep)
exten = s,n,Return()
I invoke it as:
exten = 712,1,Gosub(popcorn,s,1)
exten = 712,n,Hangup()
And lastly, attached is the source for app_waituntil.c.
It's fairly straightforward, and not very big.
But hopefully useful.
Oh, before I forget: it does require the recording of one additional
phrase,
either local or localtime. I've used local in my example
above. And
I read out the time first as GMT/UT (because I travel a lot), and then
in the
timezone of my PBX...
-Philip
___
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Re: [asterisk-users] New feature: calling all bug marshals
Joshua Colp wrote: - Original Message - From: Philip Prindeville [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Thu, 06 Dec 2007 22:34:18 -0400 Subject: Re: [asterisk-users] New feature: calling all bug marshals So, am I not summoning the bug marshals correctly? Or like druids and wizards, do they mystically sense when they are needed and emerge from the deep woods just in the nick of time? (Ever notice that whenever Batman is needed at night, it's always cloudly? How lucky is that! What if you need Batman, and it's a perfectly cloudless night? Or what if it were foggy?) Will they send me a sign if my feature gets approved? Should I look out the window towards downtown? ;-) Or... am I really supposed to file a bug after all? I think you misunderstood what the page on filing bugs said. Bugs for new features with patches are certainly welcome so open one up, fill out the license agreement, and it will get looked at. It's just feature requests without patches that we don't accept on the bug tracker. Joshua Colp Software Developer Digium, Inc. Thanks for setting me straight. *http://bugs.digium.com/view.php?id=11487 * ___ --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] New feature: calling all bug marshals
Tony Mountifield wrote: In article [EMAIL PROTECTED], Ryan Burke [EMAIL PROTECTED] wrote: I just was looking over the app_waitutil.c and am confused you add 500 to tv.tv_usec on the line msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);? It's just doing a standard round to nearest integer division, by adding half the divisor to the dividend before dividing. Without that, you just get round down instead. Cheers Tony That's right. ast_safe_sleep() has a resolution of msec, but gettimeofday() returns the time in usec, so conversion to the nearest whole msec is necessary. -Philip ___ --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] New feature: calling all bug marshals
Ira wrote: At 11:58 PM 12/4/2007, you wrote: You used to be able to dial popcorn (767-2676) in any area code (at least prior to 1982) and get the current time. I thought it was UL3-2121 when I was younger and occasionally if that was the only number in the UL3 prefix, dialing just UL3 was enough to get the time. Ira Who would have suspected that I'd be opening such a floodgate of nostalgia? :-) Anyway, can anyone tell me what other steps I might need to take to get my feature considered for future integration? Thanks, -Philip ___ --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] New feature: calling all bug marshals
Hi.
I wanted to write a popcorn app for myself, both to learn how to
script in extensions.conf, but also because it was something handy.
Along the way, I found myself doing something like:
[popcorn]
exten = s,1,Set(FUTURETIME=$[${EPOCH} + 10])
...
exten = s,n,While(${EPOCH} ${FUTURETIME})
exten = s,n,Wait(0.01)
exten = s,n,EndWhile()
exten = s,n,Play(beep)
exten = s,n,Hangup()
and hating myself for it (my Asterisk runs on a 500MHz Geode LX).
So I decided it would be useful (in general, and educational for me in
particular) to write a WaitUntil() application instead.
Well, I've done that.
I was going to file a bug and then post a fix to get their feature in,
but the Bug guidelines seem to be pretty clear that this is not the way
to go.
So, I'm posting here instead.
The example to paste into the documentation or extensions.conf is:
[popcorn]
exten = s,1,Answer()
; the amount of delay is set for English; you may need to adjust this time
; for other languages is there's no pause before the synchronizing beep.
exten = s,n,Set(FUTURETIME=$[${EPOCH} + 11])
exten = s,n,SayUnixTime(${FUTURETIME},Zulu,HNS)
exten = s,n,SayPhonetic(z)
exten = s,n,SayUnixTime(${FUTURETIME},,HNS)
exten = s,n,Playback(local)
exten = s,n,WaitUntil(${FUTURETIME})
exten = s,n,Playback(beep)
exten = s,n,Return()
I invoke it as:
exten = 712,1,Gosub(popcorn,s,1)
exten = 712,n,Hangup()
And lastly, attached is the source for app_waituntil.c.
It's fairly straightforward, and not very big.
But hopefully useful.
Oh, before I forget: it does require the recording of one additional phrase,
either local or localtime. I've used local in my example above. And
I read out the time first as GMT/UT (because I travel a lot), and then in the
timezone of my PBX...
-Philip
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2007, Redfish Solutions
*
* Philip Prindeville [EMAIL PROTECTED]
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/*! \file
*
* \brief Sleep until the given epoch
*
* \ingroup applications
*/
#include stdlib.h
#include stdio.h
#include unistd.h
#include string.h
#include errno.h
#include sys/time.h
#include asterisk.h
ASTERISK_FILE_VERSION(__FILE__, $Revision: 0 $)
#include asterisk/lock.h
#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
#include asterisk/pbx.h
#include asterisk/module.h
#include asterisk/app.h
#include asterisk/options.h
static char *tdesc = Generic WaitUntil() application;
static char *app = WaitUntil;
static char *synopsis = Wait (sleep) until the current time is the given
epoch;
static char *descrip =
WaitUntil(epoch): Waits until the current time is that given. Returns\n
immediately if the epoch is in the past.\n;
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static int waituntil_exec(struct ast_channel *chan, void *data)
{
int res = 0;
struct localuser *u;
time_t future;
struct timeval tv;
ulong msec;
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, WaitUntil requires an argument(epoch)\n);
return -1;
}
LOCAL_USER_ADD(u);
if (sscanf(data, %lu, future) != 1) {
ast_log(LOG_WARNING, WaitUntil called with non-numeric
argument\n);
LOCAL_USER_REMOVE(u);
return -1;
}
/*
* Get the current time, and calculate the number of milliseconds
* until then (rounding up from microseconds).
*/
gettimeofday(tv, NULL);
if (future = tv.tv_sec) {
ast_log(LOG_NOTICE, WaitUntil called in the past (now %lu, arg
%lu)\n, tv.tv_sec, future);
LOCAL_USER_REMOVE(u);
return 0;
}
msec = (future - tv.tv_sec) * 1000 - ((tv.tv_usec + 500) / 1000);
res = ast_safe_sleep(chan, msec);
LOCAL_USER_REMOVE(u);
return res;
}
int unload_module(void)
{
int res;
res = ast_unregister_application(app);
STANDARD_HANGUP_LOCALUSERS;
return res;
}
int load_module(void)
{
int res;
res = ast_register_application(app, waituntil_exec, synopsis, descrip);
return res;
}
char *description(void)
{
return tdesc;
}
int usecount(void)
{
int res;
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}
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Re: [asterisk-users] New feature: calling all bug marshals
Steve Edwards wrote: On Tue, 4 Dec 2007, Philip Prindeville wrote: I wanted to write a popcorn app for myself, both to learn how to script in Just out of curiosity, what does this have to do with popcorn? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 You used to be able to dial popcorn (767-2676) in any area code (at least prior to 1982) and get the current time. -Philip ___ --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 existing extensions.conf macros, and co-habitation
Anthony Francis wrote: Philip Prindeville wrote: Tilghman Lesher wrote: On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote: [snip] The issue is that I have, per virtual pbx (i.e. home or business), two contexts that these get used from. The internal-xyzzy and incoming-xyzzy contexts (one for each pbx, ie. xyzzy is home or else it's office). I was wondering if there wasn't a more flexible solution to this issue, than hard-coding a Goto(default,s,1) into them (I have no default context, because it would be meaningless). Perhaps using Gosub and Return. Or do I need to hack the macro, and pass in a 3rd argument (bletch)? MacroExit or Gosub/Return would certainly be possibilities. The main thing to note is that this macro that you call standard is actually just an arbitrary example. It is by no means perfect, so feel free to adapt it to your own liking. Sure. I just figured that it would be nice if the canned macros worked out-of-the-box without modification, in the real world. I suppose I could file a bug, and then submit patches for the macro and documentation... -Philip The ability to modify the macros to your own needs is not a bug. Anyway try adding a few more args to your stdexten to handle the context name and the like so it doesn't need default. On another point, why would asterisk come with built in code example for a multi-tenant set-up? Please save your self some time and embarrassment by not submitting that particular bug. Anthony Using the default context is a bad idea, as is pointed out in several places (including the SECURITY document, the O'Reilly book, and several good online tutorials). Besides, what's the point of having all the flexibility that you have in Asterisk if you're going to shoot yourself in the foot by having canned macros that limit that flexibility? Let's say I file the bug, and someone closes it. What's the harm? Someone doing a search of the database will at least later have the suggested patch as a possible resolution to their trying to address the same or a similar requirement. Good thing I'm not easily embarrassed, or I'd find your attitude stifling. And to address your question: it wouldn't be code *for* multi-tenant. It would be code that *didn't preclude* multi-tenant. Anything worth doing is worth doing right. Examples provided with Asterisk should showcase its power and flexibility. Not limit/ignore it. -Philip ___ --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 existing extensions.conf macros, and co-habitation
Tilghman Lesher wrote: On Thursday 29 November 2007 13:29:17 Philip Prindeville wrote: [snip] The issue is that I have, per virtual pbx (i.e. home or business), two contexts that these get used from. The internal-xyzzy and incoming-xyzzy contexts (one for each pbx, ie. xyzzy is home or else it's office). I was wondering if there wasn't a more flexible solution to this issue, than hard-coding a Goto(default,s,1) into them (I have no default context, because it would be meaningless). Perhaps using Gosub and Return. Or do I need to hack the macro, and pass in a 3rd argument (bletch)? MacroExit or Gosub/Return would certainly be possibilities. The main thing to note is that this macro that you call standard is actually just an arbitrary example. It is by no means perfect, so feel free to adapt it to your own liking. Sure. I just figured that it would be nice if the canned macros worked out-of-the-box without modification, in the real world. I suppose I could file a bug, and then submit patches for the macro and documentation... -Philip ___ --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 existing extensions.conf macros, and co-habitation
bump...
Philip Prindeville wrote:
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the default context:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press
#, return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1); Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1})
The issue is that I have, per virtual pbx (i.e. home or business), two
contexts
that these get used from. The internal-xyzzy and incoming-xyzzy contexts
(one
for each pbx, ie. xyzzy is home or else it's office).
I was wondering if there wasn't a more flexible solution to this issue, than
hard-coding a Goto(default,s,1) into them (I have no default context,
because it
would be meaningless).
Perhaps using Gosub and Return. Or do I need to hack the macro, and pass
in a
3rd argument (bletch)?
Is this doable?
Thanks,
-Philip
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[asterisk-users] Using existing extensions.conf macros, and co-habitation
I'm trying to set up my extensions.conf file using some of the existing
macros like stdexten, etc. while at the same time having two logically
separate virtual PBX's (with no default context) and two trunks coming
into separate contexts, i.e. one for residence and one for my at-home
business.
I noticed, however, that macro-stdexten depends on the default context:
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well)
; ${ARG2} - Device(s) to ring
;
exten = s,1,Dial(${ARG2},20) ; Ring the
interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1); Jump based on
status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send
to voicemail w/ unavail announce
exten = s-NOANSWER,2,Goto(default,s,1) ; If they press #,
return to start
exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to
voicemail w/ busy announce
exten = s-BUSY,2,Goto(default,s,1) ; If they press #,
return to start
exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat
anything else as no answer
exten = a,1,VoicemailMain(${ARG1})
The issue is that I have, per virtual pbx (i.e. home or business), two
contexts
that these get used from. The internal-xyzzy and incoming-xyzzy contexts
(one
for each pbx, ie. xyzzy is home or else it's office).
I was wondering if there wasn't a more flexible solution to this issue, than
hard-coding a Goto(default,s,1) into them (I have no default context, because
it
would be meaningless).
Perhaps using Gosub and Return. Or do I need to hack the macro, and pass
in a
3rd argument (bletch)?
Is this doable?
Thanks,
-Philip
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[asterisk-users] Adding new recorded phrases to the release
This might be a frequently asked question, but how do new sounds get added to the release? I was trying to do a popcorn extension on my phone (that gives the date and time... maybe even getting fancy and adjusting for the caller's timezone based on country code or area code)... but didn't have the word local or phrase local time in the lexicon. Now if I could just figure out how to grab time current time as UNIX seconds... add a small delay to it (like 5, the time it takes to sound out the time), and then wait for that time... then play a sychronizing tone... then I'll be all done: [popcorn] exten = s,1,Answer() exten = s,n,SayUnixTime(,Zulu,HNS) exten = s,n,SayPhonetic(z) exten = s,n,SayUnixTime(,,HNS) exten = s,n,Playback(vm-localtime) exten = s,n,Return() ___ --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] Urgent question.
If the hunt-group is properly done, you should be able to busy-out members of a trunk for maintenance. Otherwise, if the individual trunks have numbers (unpublished) assigned to all the circuits in the group, you could always send a Redirect() to that any of the other trunks' numbers. -Philip Alex Balashov wrote: Our provider gives us four PRIs as a trunk group hunt group. Meaning, the provider's switch will cycle through B channels in span 1, 2, 3, ... until it finds one that is available. I have moved spans 2-4 onto another machine. But we have one remaining box with a PRI full of calls and I don't know what to do with them; the box is failing, but dropping them by simply yanking the PRI is not acceptable from a business POV. Sending Congestion() or Busy() in the dial plan wouldn't work because the far-end switch would simply pass that onto the subscriber, rather interpreting it to mean that the B channel is unavailable and it should go on to other T1s in the trunk group. Any ideas? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 ___ --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] Urgent question.
Anyone have an application to robo-dial an outgoing conference call? ;-) You could tie up all your circuits with outbound calls... If you hairpin them at the switch, you shouldn't incur any usage costs... Steve Totaro wrote: To answer the question, there is currently no way to busy out a channel except to put it in use. There was some discussion about adding this feature at Astricon and on the list fairly recently. Thanks, Steve ___ --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] Help: How to configure SIP domain on SPA942
Yeah, I looked at LinksysSPATFTPProv.pdf... It doesn't say, however, how to get the phone's configuration out as a flat XML file. Only how to push the file back into the phone. Nor does it say how the phone derives its SIP domain. -Philip [EMAIL PROTECTED] wrote: Take a look at the admin guides at http://spc.pifiu.com On Nov 18, 2007 10:53 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from astlinux to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming canreinvite=no realm=redfish-solutions.com domain=redfish-solutions.com,incoming-redfish tos=184 disallow=all allow=ulaw allow=gsm localnet=192.168.10.0/255.255.255.0 externip=X.X.X.X (Footnote: do I need a default context? I'd rather not having one... I'd rather specify where my calls go explicitly...) However, my phones don't seem to be registering with any (symbolic) domain... just the IP address of their DHCP or TFTP server (can't tell which, since it's the same box). -- SIP read from 192.168.10.187:5060: REGISTER sip:192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces pbx2*CLI --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.10.187 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.10.187:5060: SIP/2.0 404 Not found (unknown domain) Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f;received=192.168.10.187 From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED];tag=as7c1c3fa2 Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 The config seems to take: Our local SIP domains: Context Set by redfish-solutions.comincoming-redfish [Configured] So, what's the DHCP option (or the HTTP knob) to tweak to get the phones to think they are in the redfish-solutions.com domain? Thanks, -Philip ___ --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 ___ --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 ___ --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] Help: How to configure SIP domain on SPA942
Johansson Olle E wrote: 19 nov 2007 kl. 04.53 skrev Philip Prindeville: I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from astlinux to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming canreinvite=no realm=redfish-solutions.com domain=redfish-solutions.com,incoming-redfish tos=184 disallow=all allow=ulaw allow=gsm localnet=192.168.10.0/255.255.255.0 externip=X.X.X.X (Footnote: do I need a default context? I'd rather not having one... I'd rather specify where my calls go explicitly...) However, my phones don't seem to be registering with any (symbolic) domain... just the IP address of their DHCP or TFTP server (can't tell which, since it's the same box). -- SIP read from 192.168.10.187:5060: REGISTER sip:192.168.10.1 SIP/2.0 It surprises me that a LInksys converts the domain to an IP address, that's broken. If you add autodomain=yes the IP address will be accepted to, or add it as a domain. The problem with these devices is that you don't know which domain they where configured for, since something is translating the domain to an IP address. With that logic, you can't separate and host multiple domains in the same SIP server. /O I don't think that's what's happening. I think it isn't having a domain (name) being explicitly set, so it's implicitly using the IP address of the TFTP or DHCP server instead. -Philip ___ --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] Trouble with asterisk-users mailman
Yeah, I posted several hours ago and I haven't seen mine either. -Philip Jesse Molina wrote: I tried re-sending my previous messages, but they are not coming through. There is definitely some kind of filtering going on with this list. I like the Report website-related issues to the webmaster. link here, which goes nowhere; http://www.asterisk.org/support/contact This is also awesome; http://www.asterisk.org/support/listrules Come on Digium -- you need to get your act together. Jesse Molina wrote: This message appears to have successfully gone through, but multiple others didn't. My messages that didn't get to the list were all sent within the first 48 hours of joining the list, but were after the first 30 minutes. I think there is something wrong. I joined the list both from my personal mail account and from my work mail account. Different ISPs. Same problem with both. Jesse Molina wrote: I'm trying this again because the last attempt didn't go through (thus more or less proving one of the below to be true.) Jesse Molina wrote: Test123 My messages to this mailing list are disappearing. Is this list quietly being moderated? Have I been wrongly black-holed? SpamAssassin gone wrong? ___ --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] Help: How to configure SIP domain on SPA942
I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone). I changed the domain in my AstLinux config from astlinux to redfish-solutions.com, and set that in my sip.conf file as well: context=incoming canreinvite=no realm=redfish-solutions.com domain=redfish-solutions.com,incoming-redfish tos=184 disallow=all allow=ulaw allow=gsm localnet=192.168.10.0/255.255.255.0 externip=X.X.X.X (Footnote: do I need a default context? I'd rather not having one... I'd rather specify where my calls go explicitly...) However, my phones don't seem to be registering with any (symbolic) domain... just the IP address of their DHCP or TFTP server (can't tell which, since it's the same box). -- SIP read from 192.168.10.187:5060: REGISTER sip:192.168.10.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:5060;expires=3600 User-Agent: Linksys/SPA942-5.1.15(a) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: replaces pbx2*CLI --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.10.187 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.10.187:5060: SIP/2.0 404 Not found (unknown domain) Via: SIP/2.0/UDP 192.168.10.187:5060;branch=z9hG4bK-1e31e66f;received=192.168.10.187 From: sip:[EMAIL PROTECTED];tag=e798d04e1a8af3a6o0 To: sip:[EMAIL PROTECTED];tag=as7c1c3fa2 Call-ID: [EMAIL PROTECTED] CSeq: 58671 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 The config seems to take: Our local SIP domains: Context Set by redfish-solutions.comincoming-redfish [Configured] So, what's the DHCP option (or the HTTP knob) to tweak to get the phones to think they are in the redfish-solutions.com domain? Thanks, -Philip ___ --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] Wanted: tutorial on troubleshooting SIP issues
Alan Lord wrote: Steve Edwards wrote: snip / Examples of what I'd like to see: 1) A SIP telephone registering successfully. 2) A SIP telephone failing to register for reasons x, y, and z. snip / I'm sorry but I don't see this as being very hard. Just install Wireshark and do it yourself... Alan You're missing the point. There's knowing *what* you're seeing, and then there's knowing *why* you're seeing it. The two are not the same. ___ --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] Wanted: tutorial on troubleshooting SIP issues
For someone that's network-aware, but hasn't sat down and plowed through umpteen SIP-related RFC's and memorized the standards, is there a good primer on troubleshooting SIP issues? I'm seeing a lot of NOTIFY/603 messages on my network between Asterisk and my Sipura 942's, for instance... Not sure what these are... perhaps the qualify keepalives? In which case, I guess the 603 is moot... but since the messages are originating from the Sipuras to Asterisk and not vice-versa, it wouldn't seem to be the qualify... Next guess would be that they're NAT keepalives, but Asterisk and the phones are on the same private subnet (which in turn *is* NATted)... Anyway, pointers for someone wanting to learn to quickly diagnose SIP conversations would be great. Thanks, -Philip ___ --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] Asterisk 1.4 from RPM
That's really a question for [EMAIL PROTECTED] The short and generally not very helpful answer is that there are a lot of poorly packaged software releases out there that don't play well with cross-development environments. -Philip Douglas Garstang wrote: I'm trying to build an Asterisk rpm from the supplied asterisk.spec file. Made numerous changes to get it to work. The architecture of the system I am building on is x86_64. I'd like to build for i686 though. I added a --target i686 to the rpmbuild line in the Makefile, but it looks like it's still requiring 64bit system libraries. When I try to install the rpm on the i686 machine, it complains it doesn't have the 64 bit libraries. How can I build with 32 bit libraries? Doug. ___ --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] Using Asterisk in SIP trunking mode with a Coppercom switch
Has anyone had any experience in getting Asterisk to interoperate with a Coppercom switch using SIP, either as subscriber lines or else as a trunked configuration? And if so, do you have any configs you could share (for both ends)? Thanks, -Philip ___ --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] Automatic provisioning of Sipura handsets (was: A linksys SPA921 behind NAT and firewall)
Erik Anderson wrote: On 10/20/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: If you are trying to use non-complied (XML) profiles... don't even bother wasting your time. Why is that? I'm using the xml-style config and they're working just fine. I'd like to be able to templatize a server, add a bunch of new handsets into sip.conf and extensions.conf, and then plug the phones into a network and have some DHCP and/or TFTP glue logic that sees the DHCP or TFTP request, and from it generates a boot file (an .XML file) and a response parameter list for DHCP... populates a file into the /tftpboot/ directory, etc. How viable is this? I'd like it to be lightweight enough that it could be done on some of the smaller embedded Asterisk boxes (like the 400MHz SoHo units). Thanks, -Philip ___ --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] Refrigerator Alarms
That a refrigerator is getting power is not the same as it operating nominally. Doors get left open... compressors fail... refrigerant eventually leaks out of seals and coils... Best to query it for temperature... and at a point faraway from the coils, such as the top of the door... which as luck would have it is also the hardest place to wire to. ;-) -Philip Kevin Withnall wrote: We use similar things here for issues like our generator battery voltage monitoring. We just have a relay going into our alarm system and as asterisk monitors our alarms it initiates emails or calls out. The alarm system is also linked into a seperate SMS unit for emergency backup so we also get SMS when any alarm goes off. My basic alarmreceiver scripts are available at http://kevin.withnall.com/2007/07/09/asterisk-alarm-receiver-using-triggers-mysql5/ if anyone wants them. -- Kevin Withnall http://kevin.withnall.com/ ILB Computing http://www.ilb.com.au http://www.ilb.com.au/ PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081 Please consider the environment before printing this e-mail ___ --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] Help 60Hz Hum?
Jay R. Ashworth wrote: On Fri, Oct 12, 2007 at 11:09:59PM +0200, F6HQZ wrote: Check if you have a ground loop. If yes, this is probably the cause of this hum. Open the loop. Actually, hum involving analog POTS lines is usually the result of the line becoming unbalanced to ground. Or else running your phone wiring in parallel with and too close to electrical (line voltage) wiring, resulting in induction (crosstalk, as it were). ___ --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
