Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4
Richard Kenner wrote: > But the question here > was *Asterisk*, not kernels. User-level code has *way* fewer > dependencies. *Precisely*. Unless we're talking DAHDI here (which we're not), Linux & ESXi are red herrings. Carlos Chavez wrote: > I am having a very tough time trying to replace an Elastix 2.X > install running as a virtual machine on ESXI 4. There's no way this has anything to do with ESXi or the version of it that you are running. Zero. Zip. Zilch. If you want to prove this to yourself and others, take the *exact* same binary bits, install them bare-metal on another piece of hardware, run the same traffic through it, and watch it crash and burn in the same way. The only way that I can see this playing out differently is if the bug (yes, bug) in Asterisk and associated libraries is extremely timing-dependent, and running it in a VM is exposing this bug in a way that most bare-metal installations wouldn't. > I will try using chan_sip > instead of PJSIP to get things running but confidence is not high. Given that the log entry you pasted into your e-mail references "libasteriskpj.so", I'd bet $$$ that switching to chan_sip has an extremely high likelihood of working, assuming that your set-up has no particular dependencies on PJSIP-specific features that you have to work around (and if you are migrating from an Asterisk 1.6 installation, I'm guessing it doesn't). Best of luck, -- Nathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Commit dialplan & other config. in memory to disk?
'lo, So yesterday, one of our clients had the misfortune of having the disk that their Asterisk config (*.conf) was stored on take a dirt nap. Of course, Asterisk was still running at the time, and everything continued to work (except for voicemail, which was stored on the same disk) right up until I shut down Asterisk to investigate what was going on. Because the disk was dead, though, I couldn't start Asterisk back up after that, and OF COURSE the backups were not firing off correctly so now we are faced with regenerating the config again (including dialplan) from scratch. In the future, if I were to ever run into a similar situation, is there any way to request or instruct Asterisk to write the current dialplan that is in memory and other important config files (e.g., users.conf) to disk in a *different* location than where it originally read them from when it started up? I could have saved myself a crap-ton of work if this were possible... Thanks, -- Nathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATA Adapter YGW30 - manual
Behold: The Wayback Machine. Link to manual: http://web.archive.org/web/20070224144946/http://www.yntx.com/files/YGW30en.rar Manual says user/pass is root/test. -- Nathan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neu pat Sent: Monday, February 13, 2017 9:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ATA Adapter YGW30 - manual I have one: YGW30 1FXS,1FXO SIP ATA unit it was made by company Yuxin I think they are no longer in business. I forgot the default user name / password for log-in. Does anybody know what was the default login or have a manual? -- Regards, Joseph -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started 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 -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to decrypt encrypted SIP user's secret
You must mean that engineer before you used "md5secret" instead of "secret" for each user in sip.conf? If so, why can't you just copy the md5secret line from the old server to the new server for each user? -- Nathan From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ikka Tirtawidjaja Sent: Tuesday, June 28, 2016 6:20 PM To: asterisk-users Subject: [asterisk-users] how to decrypt encrypted SIP user's secret Dear all, My office have an old asterisk PBX system (asterisk 11.4), and it encrypt all the SIP User's secret. But the voip engineer before me didn't save / documented those password. Now the server's hardware is begin to broke, it hangs a lot, and have a lot of call problem. We already have a new asterisk PBX to replace it, but we have difficulty to retrieve the encrypted password. about a hundred of our customer use an old IPPhone that doesn't have a reset button to hard reset the admin password (back to factory default). The previous engineer also change the IPPhone's admin password without any documentation. So, we can not move / change those IPPhone to the new PBX. Is there a way for us to retrieve / decrypt those SIP secret ? Or anyone has any experience how to reset this IP Phone (Dayou Ddip-100) Any suggestion are appreciate, because I'm really desperate. Thanks in advance, Regards, Ikka Jakarta - Indonesia -- _ -- 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] Calling multiple phones at ones
On Monday, June 15, 2015 at 9:03 AM, Matthew Jordan wrote: > This is true for chan_sip. It is not true for the PJSIP stack. > > The PJSIP stack does allow for multiple devices to register contacts > to a single Address of Record (AoR). You can then dial contacts > individually, or dial all contacts on an AoR using the > PJSIP_DIAL_CONTACTS function. Very interesting and good to know; thanks. I'll have to check it out when I've got some spare time. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling multiple phones at ones
On Monday, June 15, 2015 at 9:21 AM, Nilesh Govindrajan wrote: > How about ringall strategy with a queue? Not sure how that would help. Every SIP phone in the queue would still have to have a unique SIP identifier/username. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling multiple phones at ones
What you want is called SIP call forking, and unfortunately, last time I checked (before Asterisk 12 and the advent of PJSIP), Asterisk's SIP channel driver does not support it, and I would be shocked if Asterisk 12+ changes this situation. You can even see that people have written and submitted patches for this in the past, but they have been rejected: https://issues.asterisk.org/jira/browse/ASTERISK-13614 It has apparently been a somewhat contentious issue. Asterisk's philosophy is that it is not a SIP proxy, but a multiprotocol PBX that also happens to support SIP endpoints, and so the channel drivers need to be as generic as possible and anything that can be passed on to the dialplan to be handled in a uniform and consistent fashion should be, and that would include call forking. The developers do not want for there to be two places (a universal way and a channel-specific way) where this kind of functionality can be configured. You *can* fork calls with Asterisk; don't get me wrong. Simply specify multiple endpoints to ring when you execute the Dial() application, concatenating them together with ampersands, like so: [office-phones] exten=555,1,Dial(SIP/555&DAHDI/g0/5551212&SIP/567) As you can see, you can mix-and-match channel technologies in this way. In the above example, when a call for extension 555 is received within the office-phones context, the SIP user 555 is called, the SIP user 567 is called, and the PSTN phone number 555-1212 is called, all simultaneously. The first phone to answer the call gets it. The problem that I have with this method is that if a particular channel technology or protocol has a mechanism specifically for performing call forking defined within the standard/spec for that protocol (as SIP does), there is no way with Asterisk to take advantage of the channel-specific way to accomplish this. Let me unpack this a bit more. As you can see, with the above example, if your published extension number is 555, and you want to have two different SIP phones ring at the same time when you get a call, you can't actually register them both to your Asterisk server as user '555'; they each need to have unique SIP usernames. So you have one phone that is registered as 555, another one that is registered with a throwaway extension number (567), and then you instruct Asterisk to send the call to both 555 and 567. Because Asterisk's SIP implementation does not support native SIP call forking and also assumes unique usernames for every SIP peer, if you try to register two SIP endpoints with the same username (e.g., 555), the second registration will actually be successful, but it will *overwrite* the registration for the first phone in memory, meaning that the phone that first registered with that username will stop receiving calls and only the second phone will get those calls. And as the SIP registration timer expires for each phone and they refresh their registrations with the server, they will be constantly overwriting each other's registrations, so incoming calls will constantly switch between which phone is getting the calls. It's a mess. It's not that a single SIP registrar or proxy cannot have multiple endpoints registered to it with the same username; this is actually specifically allowed by the spec and many SIP-only proxies support this. It's just that Asterisk does not, which means that you have to resort to (IMO) ugly hacks like creating a bunch of unpublished extension numbers for your additional phones. I completely understand where the developers are coming from, but it would be great if this tension between the Asterisk philosophy and the need for this feature could be resolved via some sort of compromise, like, for example, allowing for the SIP channel driver to accept and track multiple registrations for the same SIP user, and then exposing each of these registrations as separate entries in the peers table that can be individually addressed within the dialplan. Or something. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Demkovitch Sent: Sunday, June 14, 2015 7:13 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Calling multiple phones at ones Hello group! I’m new to Asterisk but got one running finally :) Now I’m trying to solve following problem. I have company Automated Attendant and each employee have SIP phone at home, SIP phone in office, cell phone. I want all those 3 phones to be “one”. So, if someone calls our company number and dials my extension - I’d like 3 phones to ring at the same time. What is this feature and where should I look for samples, etc? I’m going by “Asterisk: The definite guide” book and pretty confident with those concepts described b
Re: [asterisk-users] Polycom DND + Intercom/Paging Override?
On Thursday, September 18, 2014 10:31 AM, John Kiniston wrote: > There is one product that I know of that is Compatible with Polycom > paging. The Algo 8180 Audio Alerter. [snip] > > You can call it via SIP from asterisk and it can multicast in the special > Polycom format to your phones. Wow, I had no idea! I have looked at SIP-based PAs in the past, including this one, but this completely escaped my attention. I just browsed through the manual, and sure enough, this is an advertised feature. Kinda weird that you have to buy an all-in-one loudspeaker to acquire a device that can act as a SIP-to-Polycom-multicast bridge...it would be nice if they sold a cheaper version that omitted the speaker. (Or, even better yet, if Asterisk just supported this natively so that you didn't have to buy some hardware box.) But still, it's nice to know that this exists and is an option. Thanks for the heads-up! -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom DND + Intercom/Paging Override?
Yes, I am pretty sure that if a Polycom unit is set DND and you initiate a multicast page from another Polycom handset on a page or PTT channel that the DND handset is subscribed to (like the emergency channel), then you will hear audio on that handset. BUT Polycom handsets cannot be configured to just listen to RTP being multicasted to a particular multicast IP like many other IP phones can...the signalling for Polycom multicast paging and PTT functionality is completely proprietary and not SIP-based, and in fact the audio itself is not RTP. It is a proprietary audio packet format that has a header prefixed to it containing signalling information, on every audio packet/frame. Therefore nothing else can initiate a multicast page except another Polycom phone on the same layer 2 broadcast domain...you cannot programmatically have Asterisk/FreePBX do this. Polycom has released an engineering advisory documenting the format, in case anyone in Asterisk land is interested in writing a channel driver that can interoperate with this. I for one think it would be very handy to be able to have Asterisk initiate group paging and push-to-talk on Polycom handsets. The document is here: http://support.polycom.com/global/documents/support/technical/products/voice/Audio_Packet_Format.pdf -- Nathan Anderson First Step Internet, LLC nath...@fsr.com On Wednesday, September 17, 2014 6:03 PM, David Wessell <> wrote: > Tim, > > I THINK but I'm not sure that you can do this with the Polycom multicast > page function. Have you attempted this yet? > > Thanks > david > > On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson > wrote: > > > Greetings- > > As many of your are Polycom "experienced", I was hoping some kind soul > could provide direction on a specific issue. > > On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding > an instance where, using intercom/paging functionality of FreePBX, I need > to override an end user's 'Do Not Disturb' selection on the handset. By > default, DND simply rejects all inbound SIP INVITEs. However, a > page/intercom needs to be allowed through. > > Any suggestions? I've read reports this is doable using Polycom config > options for call priorities, but I've had no such luck yet. > > Thanks! > > --Tim > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to provision asterisk's phonebook to Polycom vVX310's
On Tuesday, January 21, 2014 12:32 PM, Adrian Serafini wrote: > On 01/21/2014 01:55 PM, Stanley van Dijk wrote: >> Hi, >> Am running a freepbx install and created trunks, extensions and groups. >> Now I'd like to hand out the Asterisk phonebook to the phones (all VVX >> 310's). Is there an easy way to do this? > > Even the old ones could view a webpage. Have a script read the Mysql > DB/users table data, then output in XML. The newer ones can output in > HTML5. This solution is auto updated when you add GUI users. ...or, if you use res_phoneprov, Asterisk can auto-generate a "static" XML phone directory from a template during run-time (whenever the file is requested). I'm not sure how one would go about setting this up if they are using FreePBX, however...I use res_phoneprov in combination with Asterisk-GUI and it works great. > Polycom used to charge for LDAP directory access, Pretty sure you're correct, and that this hasn't changed (AFAIK). -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qualify=yes: OPTIONS: How to Change?: `From: "asterisk"`
On Thursday, May 09, 2013 8:23 PM, Jeremy Kister wrote: > On 5/9/13 8:21 PM, Brian LaVallee wrote: >> When qualify is enabled on a trunk, the From line shows "asterisk". See >> the SIP message below. > > I had the same annoyance/issue. fixed it in > https://issues.asterisk.org/jira/browse/ASTERISK-17616 > > the patch was included in 1.8.9 rc1. Interesting. I hadn't noticed this bug or its inclusion into 1.8.x. IIRC, pretty sure I worked around this myself in the past by setting a global "callerid=" value in sip.conf, so if you have a good (!) reason not to upgrade, the OP might give that a shot. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Radius Based Accounting for Asterisk
On Friday, April 26, 2013 11:38 PM, basteon wrote: > On 3 February 2011 09:44, Nikhil wrote: > > > Hi everyone > > Any one used Radius based accounting for asterisk.Please give me > > details. > > hi, > you still interesting in it? > that I made long time ago. > http://lists.digium.com/pipermail/asterisk-dev/2010-March/043096.html > also I keep another patches and things and I need dedicated ftp for > it. if you can give me such things I'll provide this patch to you. I have to say, I'm confused. Asterisk already supports RADIUS Accounting (for CDRs), and has for quite some time. What it doesn't support is RADIUS Authentication. I'd have to think that even if it did support RADIUS Auth, barring some kind of architectural overhaul, it would have to happen on a channel-driver-by-channel-driver basis. I, for one, would love to see true RADIUS Auth show up in chan_sip at the very least. As I understand it, there are a couple of obstacles inherent in the current architecture that makes this a not entirely simple problem to solve. The first obstacle is that chan_sip currently needs to know each peer's secret, whereas if you are using a RADIUS server, the actual secret/password would never be known to Asterisk. SIP uses HTTP-style digest authentication, which can be tunneled over RADIUS, but chan_sip would itself need to be rewritten to be "RADIUS-aware", so that rather than require/expect the secret for a peer to be statically set in sip.conf and then perform the digest auth exchange itself, it simply sends the request on to the RADIUS server. There was an effort some time ago (https://issues.asterisk.org/jira/browse/ASTERISK-5278) to separate peer authentication into a separate module that would have a plug-in architecture which would allow you to "bolt-on" different kinds of authentication back-ends as needed (RADIUS, PAM, LDAP, etc.) for channel drivers that supported it, but development of it was put on-hold back in 2007 and, as far as I can tell, hasn't been picked up by anyone since. Even if res_auth had been finished and was in a working state, the other obstacle that nobody ever really attempted to address with it (as far as I was able to tell, from reading through the bug) was that even though it solved the secret-must-be-known-by-Asterisk problem, you still had to statically define all of your SIP (or IAX2, or whatever) peers in the appropriate conf file. You would merely add an option that told it to check the password/secret via RADIUS, or PAM, or what-have-you, for that peer instead of statically typing the plain-text secret in the conf file for that peer. If you still have to define all of the peers themselves in sip.conf, then doesn't that essentially make the RADIUS "support" useless? What I suspect most people who want to use RADIUS with Asterisk would *expect* from such support is the ability to use RADIUS as the authoritative source for all peer definitions, and not merely as a central password store. That is, they want to be able to utilize all 3 of the three-As of RADIUS AAA: Authentication, Authorization, and Accounting. res_auth would have only taken care of authentication, not authorization (peer attributes, permissions, and settings). You *can* create "dynamic" peers in chan_sip and chan_iax2 by way of Asterisk's Realtime Architecture. And so I could forsee someone cobbling together RADIUS support by resurrecting res_auth (for authentication), and then perhaps creating a Realtime module (for authorization) that consulted a RADIUS server instead of a SQL database or whatever. But you'd have to use both in conjunction with each other for a fully-working system: the Realtime RADIUS module would have to add the appropriate option to each SIP (or IAX2) peer that would in turn cause chan_sip to consult the RADIUS server for that peer's password, instead of cacheing the password/secret locally. And that just seems real kludgey to me (although it might be better than nothing!). -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones
On Apr 28, 2013, at 13:56, "Carlos Alvarez" wrote: > If the SIP peer exists, they simply fail silently, with no error in the CLI > or the messages log. Nothing works, but no errors. Maybe 'sip set debug peer xxx' where 'xxx' is the peer name, and then try to see if you can spot what it's doing wrong/differently during registration vs. working phones. -- Nathan -- _ -- 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] E911 Voip Trunking
On Friday, April 19, 2013 5:35 PM, Warren Selby wrote: > On Fri, Apr 19, 2013 at 10:41 AM, Chris Nighswonger wrote: > > > During the course of a conversation with an member of the IT group who > > handles the E911 center for our county, I learned that all of the > > county's E911 is voip based. This got me to wondering why we could not > > just configure up a SIP or some such trunk directly to the E911 center to > > handle our emergency traffic. The county seems interested in exploring > > the possibility. > > There are E911 providers that offer this functionality. I know off the > top of my head, 911Enable offers a service like this. A former client of > mine that provided hosted PBX services had a contract with them. I'm > sure there are other providers out there as well. Indeed. 911ETC is who we use, and is another example. Even if you could peer directly with your county's PSAP, in the case of 911, I think it is a way better idea to go with one of these specialty SIP-based E911 providers, for the simple reason that even if you only sell VoIP service to people residing within your county, ATAs and VoIP phones are nomadic in nature: people are going to take them with them when traveling/on vacations, or maybe even use a soft phone with their account. This means that they are going to need to have the ability to update their physical E911 location, so that when they are away from home or away from the office, their 911 calls are directed to the correct local PSAP for their current location, and not back to their home county's PSAP. So, sure, you might be able to convince your county PSAP to peer with you directly via SIP, but it's not realistic to then go out and do the same for the other 8,000+ PSAPs in the U.S. (http://transition.fcc.gov/pshs/services/911-services/enhanced911/psapregistry.html) that one of your customers *might* be closest to at any given time, not to mention purchase and maintain the infrastructure, technology, and data needed to accurately geocode a physical address and then map it to a given PSAP. That's what these services are for: they deal with all of that, and all you have to do is send 911 calls to their SIP proxy, and they route it appropriately. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic realtime + queues.conf Unresolved
On Friday, April 19, 2013 1:45 PM, Tommy Cooper wrote: > The following error suggests that my syntax is incorrect, that syntax > seems to be part of an SQL query. I do not have any SQL queries anywhere > within my configuration. iODBC or unixODBC? I'm sure that the query is being generated on-the-fly by res_config_odbc based on information it is being fed by app_queue; there is no .conf file that you can edit to see or modify the query. If Asterisk is doing something wrong when generating this query, it'll most likely have to be addressed in source somewhere. It would definitely be helpful to see what the *whole* query is that Asterisk is trying to execute. Try to enable trace/logging for your ODBC driver. If unixODBC, for example, add: [ODBC] Trace = Yes TraceFile = /tmp/sql.log ...to your odbcinst.ini file. This will dump detailed diagnostic information to the TraceFile, including the actual queries (in full) that are being executed. Just from what little we already know from the Asterisk logs, though, it *almost* looks like an escaping problem of some kind. What do you have the 'backslash_is_escape' option for your DSN set to in res_odbc.conf? Maybe try setting it to the opposite of whatever it's configured for now. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer only, no outbound calling
On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote:
> New Problem, now operators can pick up the previous inbound only line and
> dial out to anything that matches the patterns I have defined in the
> context for their extension in sip.conf.
>
> What I really need to make work here is Attended-Transfer since that is
> what is desired by those using the system.
I'll assume we are talking about SIP extensions here.
What is doing the actual transfer? Is it Asterisk (res_features /
features.conf), or the phones themselves?
If it is the phones themselves, you're probably out of luck because in an
attended transfer scenario, the transferor has to send a regular ol' INVITE to
the transfer target before sending a REFER to the transferee, and so there's
really no way that Asterisk can know whether that INVITE to the transfer target
is someone in the middle of attempting an attended transfer, or someone trying
to place a regular outbound call. Your only hope would be to sniff the SIP
traffic between your handsets and Asterisk, and see if there is a SIP header
difference that is detectable between what your phones generate for an attended
transfer vs. an outbound call. If there is, you can use the ${SIP_HEADER()}
function in your dialplan to check for the presence of that difference in order
to determine whether a call is an attended transfer or not.
If you have the option of using Asterisk's built-in attended transfer feature
(features.conf + passing option 't' to the Dial() command that calls a given
extension for an inbound call) instead of a button on your phones, you can
override which context a transfer target's number is executed in by overriding
the global variable TRANSFER_CONTEXT. So you can create a new stub context
that sets your variable to let you know that this is a transfer and then jumps
to the SIP client's normal context, and set TRANSFER_CONTEXT=your_new_context
under the [globals] section of extensions.conf. Check for the presence of your
variable in the SIP client's context, and act accordingly.
Note that in either scenario, as long as you allow attended transfers, the
system can be gamed by people. For example, assuming that extensions can call
other extensions, someone who wants to make an unsanctioned outbound call
simply walks over to a vacant phone in another cubicle, calls their own
phone/extension, rushes back to answer it, and then initiates an attended
transfer that they never end up completing (they just talk to the person they
initiated the transfer to the whole time).
Hope this helps,
--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com
--
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Re: [asterisk-users] [OpenSIPS-Users] 404 When BYE initiated by external callee
On Tuesday, April 09, 2013 1:31 PM, Nick Khamis wrote: > As I see asterisk rewrites the callid unexpectedly when initiating the > INVITE with the SIP trunk (trace packet 4). [...] > Asterisk has mapped the call with the two different ids together. Nick, As Joshua has already tried to explain to you, Asterisk is a B2BUA, not a SIP proxy. This is by virtue of its nature and origins as a technology-agnostic PBX. It is not "rewriting" the Call-ID. It's generating an entirely new one because the INVITE that it generates is considered by Asterisk to be a competely new/separate call leg. It then maintains a table of which call legs are "bridged" together into a single call, regardless of the underlying channel technology. Because of how Asterisk works under-the-hood, it is also impossible for it to "pass on" Record-Route header fields to the other leg of the call. It will, however, take appropriate action in passing any signalling events downstream (for example, in your case, a "BYE" will be sent to one call leg if it is received on the other, but NOT because it is proxying it; to boil it down, internally, the received "BYE" is translated to a generic "hang-up" event which the SIP channel driver takes and uses to generate a completely new "BYE" from scratch on the other leg). I concur with Joshua: this is not an Asterisk problem, and what it is doing is completely reasonable. I suspect that the "leads" you are chasing in this investigation will turn out to be a red herring. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Pattern matching repeating digits
Eric,
Thanks; of course, this is also an option. However, setting up a separate
context for this type of thing with several identical Goto statements also
strikes me as inelegant, even if it is less so.
-- Nathan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Thursday, March 28, 2013 8:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pattern matching repeating digits
You are correct, it is stupid 8-)
exten => 233,1,Goto(dial-out,${EXTEN},1)
exten => 255,1,Goto(dial-out,${EXTEN},1)
[dial-out]
exten => _XXX,1,DoStuff()
exten => _XXX,n,AndMoreStuff()
exten => _XXX,n,Dial(something)
exten => _XXX,n,Hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nathan Anderson
Sent: Wednesday, March 27, 2013 2:18 AM
To: 'asterisk-users@lists.digium.com'
Subject: [asterisk-users] Pattern matching repeating digits
'lo, all,
Is there some (possibly undocumented?) way that I can pattern-match on a
specified number of repeating digits? (Something similar to regular
expressions' {})
Here's an example: let's say I have a string of things that need to be done for
both extensions 233 and 255. I can either...
A) Repeat the exact same code for both extensions, like so:
exten => 233,1,DoStuff()
exten => 233,n,AndMoreStuff()
exten => 233,n,Dial(something)
exten => 255,1,DoStuff()
exten => 255,n,AndMoreStuff()
exten => 255,n,Dial(something)
...which is stupid, or...
B) I can attempt code reuse for similar cases (a Good Thing[tm]), and make as
specific of a match as possible, like so:
exten => _2[35][35],1,DoStuff()
exten => _2[35][35],n,AndMoreStuff()
exten => _2[35][35],n,Dial(something)
...but this will not only match 233 and 255, but 235 and 253 as well.
It'd be nice if there was a substitute character that meant "a character that
is exactly the same as the preceding one"; for example, if R was meant to
represent such a concept, then this would do what I want:
exten => _2[35]R,1,DoStuff()
exten => _2[35]R,n,AndMoreStuff()
exten => _2[35]R,n,Dial(something)
You could even do crazy things like chain them together (this would match 2
and 2 and nothing else);
exten => _2[35]RRR,1,DoStuff()
exten => _2[35]RRR,n,AndMoreStuff()
exten => _2[35]RRR,n,Dial(something)
Am I missing something or does this really not exist?
Thanks,
--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com
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[asterisk-users] Pattern matching repeating digits
'lo, all,
Is there some (possibly undocumented?) way that I can pattern-match on a
specified number of repeating digits? (Something similar to regular
expressions' {})
Here's an example: let's say I have a string of things that need to be done for
both extensions 233 and 255. I can either...
A) Repeat the exact same code for both extensions, like so:
exten => 233,1,DoStuff()
exten => 233,n,AndMoreStuff()
exten => 233,n,Dial(something)
exten => 255,1,DoStuff()
exten => 255,n,AndMoreStuff()
exten => 255,n,Dial(something)
...which is stupid, or...
B) I can attempt code reuse for similar cases (a Good Thing[tm]), and make as
specific of a match as possible, like so:
exten => _2[35][35],1,DoStuff()
exten => _2[35][35],n,AndMoreStuff()
exten => _2[35][35],n,Dial(something)
...but this will not only match 233 and 255, but 235 and 253 as well.
It'd be nice if there was a substitute character that meant "a character that
is exactly the same as the preceding one"; for example, if R was meant to
represent such a concept, then this would do what I want:
exten => _2[35]R,1,DoStuff()
exten => _2[35]R,n,AndMoreStuff()
exten => _2[35]R,n,Dial(something)
You could even do crazy things like chain them together (this would match 2
and 2 and nothing else);
exten => _2[35]RRR,1,DoStuff()
exten => _2[35]RRR,n,AndMoreStuff()
exten => _2[35]RRR,n,Dial(something)
Am I missing something or does this really not exist?
Thanks,
--
Nathan Anderson
First Step Internet, LLC
nath...@fsr.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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