[asterisk-users] sip.conf, realtime, and LDAP
I'm confused exactly what's supported with LDAP and Asterisk. What I want to do is to have SIP peer information read directly (in realtime) from LDAP. Can this be done? If so, with what Asterisk versions? -- _ -- 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] Top Posting
It is not a matter of preference, it is actually a rule [1]. Top-posting is also an annoying practice [2] and NOT the general accepted way to reply. And that's been the case for at least TWO DECADES. I find it amazing that this is still being argued now. -- _ -- 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] extconfig, realtime, and SIP
I'm confused about a few things relating to realtime, SIP and config in general. As I understand it, with the exception of extensions.conf, I can either have a config file completely in text or completely in a database. Is that correct? I can't find documentation for exactly what switch = does but is that only in the dialplan and a way to have it partly from a file and partly from a database? For SIP, do I understand it correctly that I can have sip tables both via realtime AND in sip.conf? For the LDAP realtime, how can I implement setvar? -- _ -- 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] Hide the plain text password
Who are you hiding them from? Anyone with access to the Asterisk server can already do far more damage than extracting these passwords. You may (like we do) want to store config files in a version control system in a common repository. People who have access to that repository don't necessary have access to the Asterisk server. -- _ -- 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] Hide the plain text password
Anyway, the answer is: No, it's mathematically impossible to do that. Even if the passwords were stored encrypted, Asterisk itself has to be able to get the plaintext passwords to send to the remote server; so the code to decrypt them must necessarily be located on the machine. And the Source Code to Asterisk is readily available, which is how come you were able to benefit from it, so it would be trivial to extract the passwords in any case. But there IS a way to improve things, and it's what Cisco routers do. You can have all password stored in config file encrypted with a single master key. That key is stored in a special file, containing just that key. THAT file must then be heavily-protected, but all OTHER config files can now be placed into CM or anywhere else they might be needed. -- _ -- 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] Hide the plain text password
How does that improve things? The reason that works with Cisco routers is because the code that reads that special key file and uses it to decrypt the other files is closed-source; nobody can see how it works. As another poster said, that's not true for Asterisk. If Asterisk had such a facility, the method used to decrypt the protected passwords would be publicly available, as would the decryption key (in the special key file). Anyone who wanted to decrypt the passwords from the config files would have an only slightly more complex route to do so... it would still be straightforward. Please reread what I wrote. The encryption key for the passwords wouldn't be in Asterisk sources, but selected BY THE USER and stored in a SINGLE configuration file that contains just that password. This is what Cisco does. That way, the rest of the config files, which you might want to put in a CM system, need not be protected. -- _ -- 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] Hide the plain text password
Right. But it really won't help much (except complicating things) if the user has decent access to Asterisk. Yes, but we're talking about cases where the user *doesn't* have access to Asterisk. At many locations, including mine, Asterisk runs on a machine dedicated for that purpose and only people administering it have access to that machine. But config files are placed in a CM system which MANY more people have access to. Having plaintext passwords in those files is a real problem. -- _ -- 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] Hide the plain text password
#include the password (a file the line 'secret=') from a local file on
the file system. The user has no access to it, right?
Right, but we're not talking ONE password, but ANY password. Having
dozens of those files, one for each password, gets to be a real pain
really fast. And you STILL want CM control of password changes even
if you're storing the encrypted versions: you want to be able to go
back to an old password, even if you don't know what it is.
One test for you to consider: are the users able to use the encrypted
configuration item in a different Asterisk system (without your
concent)?
Of course not! It would be useless if that were the case: the whole
point here would be that you need the master encryption key.
Here's a possible design:
- There's optionally a file in the config
directory called master_key. It contains just a string.
- A CLI command core encrypt string is added to Asterisk. It takes the
provided string, encrypts it using the string in master_key, and outputs
a string of the form {enc:encrypted_version_of_string}.
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
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Re: [asterisk-users] Hide the plain text password
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
This sounds pretty reasonable, except perhaps that you might only want
to convert strings in password fields -- otherwise you risk false
positives in e.g. the dial plan.
I think this works much better if it's purely lexical. Otherwise, you
have to teach the code what's a password and what's not and maintaning
that is an ongoing issue, so I think a cleaner design would be to pick
some string that's just not going to occur anywhere.
I can recommend contracting with one of the indepedent Asterisk
developers to get this done. You will likely find them on the
Asterisk-biz-list.
I could easily do it myself if it were something that I personally needed
(except that I'm not sure if two-way encryption routines already exist
in Asterisk), but we don't have enough passwords for this to be an issue.
I was posting the design to address the issues raised by the person who
started the thread.
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)
I recognize all the options given yet as I explained before they are not viable. I do not have the resources to pay someone, I do not have the expertise to fix this issue because according to an asterisk developer any fix in that area would be deeply architectural in nature... what other options are there? In a commercial product, you have two options when you find a bug: (1) Pay for it to be fixed. (2) Live with the bug. In an open-sourced product, you have those same two options, plus an additional one: (3) Fix it yourself. Those are the only three you have to choose from. -- _ -- 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] [1.4] Reading phone number the French way?
exten = s,n(nbr2call),Read(NBR2CALL,please-type-number,10,,2,20)
For instance, a landline number in Paris like 01 42 92 81 00 is read
zero-one, forty-two, ninety-two, eighty-one, zero-zero, where I
assume Americans would read all the digits individually (zero, one,
four, two, etc.)
Maybe something like:
exten = s,n,SayDigits(${NBR2CALL:0:1})
exten = s,n,SayNumber(${NBR2CALL:2:2})
exten = s,n,SayNumber(${NBR2CALL:4:2})
exten = s,n,SayNumber(${NBR2CALL:6:2})
exten = s,n,SayNumber(${NBR2CALL:8:2})
Or make changes in say.conf.
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Re: [asterisk-users] The new ConfBridge application is now in Asterisk Trunk!
No, conference scheduling is not a feature that we have built directly into ConfBridge, and I'm debating on what it would look like. Scheduling isn't built into MeetMe either, but the fact that it dynamically reads from a database means that you can write external programs (such as Web-Meetme) that create conferences that MeetMe can read. For me, in order for ConfBridge to be at all interesting, it needs the same functionality. -- _ -- 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] Odd error in libpri
I just updated libpri 1.4 on my system to the latest from that branch and my QSIG connection to an NEC SV8300 stopped working. The trace showing the problem is below: q931.c:5640 q931_connect: Call 7168 enters state 10 (Active). Hold state: Idle DL-DATA request Protocol Discriminator: Q.931 (8) len=21 TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) Message Type: CONNECT (7) TEI=0 Transmitting N(S)=29, window is open V(A)=29 K=7 Protocol Discriminator: Q.931 (8) len=21 TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) Message Type: CONNECT (7) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: NET] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] [29 05 0b 05 01 0e 03] Time Date (len= 7) [ 11-05-01 14:03 ] Protocol Discriminator: Q.931 (8) len=13 TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent from originator) Message Type: STATUS (125) [08 03 81 e0 29] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 29 (41) [14 01 04] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Delivered (4) Received message for call 0x2aaab81d15c0 on link 0x1b0db440 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) As I'm reading this, libpri thinks that the SV8300 is complaining that a mandatory IE is missing, in this case time/date. However, the field is THERE. But when I go back to a working libpri (r1878), I see that the time/date is NOT sent on the CONNECT. If I'm reading Q.931 correctly, 5.1.8 (page 118) says that the Date/time IE may be included as a network option. I see this was added to libpri at revision 2187, in response to issue number 18047. I played around a bit. Since the spec includes seconds, I added seconds to see if that made it work, but it didn't. I DID work when I deleted Q931_IE_TIME_DATE from connect_net_ies. Whether or not it's a bug for the SV8300 to reject that IE, it's likely that NEC won't fix it. This likely means that a new config option is needed, but I think that means it'd also have to be done in chan_dahdi.c in Asterisk in addition to libpri. Is that right? -- _ -- 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] Odd error in libpri
Please create a mantis issue describing this problem. Pardon my ignorance, but what does mantis refer to? -- _ -- 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] Issue with Asterisk Aastra 57i at v3.2
I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says contact mismatch. I added sip contact matching: 2 to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the Aastra sending a REGISTER and Asterisk replying with the 401. The phone then sends the REGISTER again, this time with the hash. Asterisk now replies OK, but sends an OPTION packet FIRST and I think that confuses the Aastra. Has anybody seen this? Is there any way to have the packets sent in the proper order? -- _ -- 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 Asterisk Aastra 57i at v3.2
Is asterisk replying differently when firmware 3.2 is used ? No, but the phone cares with 3.2 and not with 2.6. -- _ -- 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 Asterisk Aastra 57i at v3.2
Asterisk does indeed send an Options before the OK but my 57i doesn't seem to mind. That's odd. It does for me. Perhaps you need to upgrade firmware on the Aastra phone? The problem occured when I DID upgrade it! Precisely to the one you mentioned. Or turning off qualify for this peer might work-around it for you. I'm sure it would, but all peers are those phones, so that's not an acceptable workaround. -- _ -- 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 Asterisk Aastra 57i at v3.2
In that case it suggests it is some setting you have applied to the phones that is causing it. Can you post the local.cfg server.cfg files from the phone (removing the passwords from there first)? Sure: local.cfg is checksums, server information, and: contrast level: 3 ringer volume: 8 contact rcs: 0 sip line1 dnd: 1 Server.cfg removing server/account/password/button/softkey information is below. Note that I added the contact matching config when I ran into this problem, so it failed with and without it. # options simple menu: 1 time server disabled: 0 time server1: pool.ntp.org time server2: time.windows.com download protocol: HTTP web interface enabled: 1 auto resync mode: 3 auto resync time: 03:20 emergency dial plan: 911 sip dial plan: [12367]xx|5xxx|*xxx|#[1-9]|9;1xx|9;[2-9]x|9;011x+#|8;x+#|#0[12367]xx|#05xxx|#0*xxx|#0#[1-9]|#09;1xx|#09;[2-9]x|#09;011x+#|#08;x+#x sip proxy ip: gnat.com sip contact matching: 2 sip rport: 1 sip update callerid: 1 sip blf subscription period: 120 sip explicit mwi subscription period: 120 sip accept out of order requests: 1 sip use basic codecs: 1 sip silence suppression: 0 sip vmail: #6 live dialpad: 1 directory 1: main.csv directory 2: special.csv dnd key mode: 1 call forward key mode: 1 call hold reminder: 1 call hold reminder timer:60 call hold reminder frequency:60 call waiting:1 call waiting tone period:5 bl on time: 600 https validate certificates: 0 xml get timeout 10 preferred line: 1 softkey selection list: none, speeddial, blf, dnd, callforward sip xml notify event: 1 time zone name: US-Eastern But it might allow your users to make phone calls while you fix the issue properly. Calls can be MADE with the new firmware, just not recieved. A better workaround for me is to just stay with the old firmware. -- _ -- 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 Asterisk Aastra 57i at v3.2
In that case it suggests it is some setting you have applied to the phones that is causing it. I just called Aastra tech support. I'm always VERY impressed that the first person who picks up the phone is very technical. He said that they've had reports of this issue. The problem goes away when sip rport: 1 is removed. Even though the manual says that it's recommended for NAT, it isn't needed and is causing this problem. They appear to be treating this as a bug. (There's supposed to be a new version of the firmware very soon, since it's ready, but just not on the web, but it won't solve this issue.) -- _ -- 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] receive faxes
I don't believe you really understand what Open Source means...it does not mean FREE. Actually, it DOES mean free, especially since Asterisk is under the GPL. But, as RMS often says, that's 'free' as in 'free speech', not 'free beer'. That problem doesn't exist in French, where there are two distinct words for free and they refer to it as libre. The point of Free (or Open Source) Software is that if you think that some company (e.g., Digium) is charging too much to maintain it, you have the freedom to maintain it yourself of even start your own company and compete with them. -- _ -- 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] Free CNAM
FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 I just tried this on about a dozen numbers I have in various parts of the US (cell, business, and a landline number I've had for decades) and NONE of them were listed in this database. Indeed I can't find one that IS. -- _ -- 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] Free CNAM
Try them all again. Remember that this is a static database that has to 'research' numbers it has not seen before. Well, that doesn't make it very interesting: most calls I'd expect to get won't have been seen by it before. By now (a few minutes later), the database should have been updated. I tried it, but it returns the same kind of junk that some of the databases do. For example, on a Florida number, it just says FLORIDA instead of the proper name (some of the CNAM databases have the right name). -- _ -- 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] Free CNAM
Try them all again. Remember that this is a static database that has to 'research' numbers it has not seen before. What happens when the CNAM is changed? How often does it go back and poll the database? -- _ -- 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] Free CNAM
The system uses real Telco CNAM Dips. Any generic names you get are from the subscriber's carrier itself. We can only provide what we ourselves get. There's more than one CNAM database (aren't there seven?). I would have hoped that a service such as this would look at a bunch of them and choose the one that had the best result. -- _ -- 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] Get second cipher in an extension
how can I get the second character/cipher of an extension ?
If I have : exten = 12345,n,NoOP()
How can I get 2 ?
${EXTEN:1:1}
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Re: [asterisk-users] Securing Asterisk
Can please the Powers that Be reconsider and add this option to sip.conf? What Powers that Be? This is open-source software! If you need an option in sip.conf, just add it! -- _ -- 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] Licensing question.
But so long as you were careful not to copy any of the code you are going to link against into your Source Code (and why would you, if you were linking against it?), it only *becomes* a derivative work *after* it has been compiled. That's not necessarily true because if you have a work that cannot be used independently (e.g. a plug-in), there are numerous court precedents that say that it indeed is a derived work. This area of the law is very complex and people should really consult an attorney experienced in this area if they care about such things. -- _ -- 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] Clipping issue with SIP over satellite
I'm having a wierd clipping issue with one employee who's using a phone over a satellite Internet. He was sold that system specifically for use with VoIP. Ping times show average round-trip time as around 700 ms with a range of 560 to 841, so considerable jitter. Things work fine when he's talking to another Asterisk phone or to a SIP trunk provider, but when connecting to a T1, there's clipping where about 1/3 of his voice (in intervals of maybe 200ms) are removed. This sounds like an echo canceller conflict, but I've set echocancel=no in chan_dahdi.conf (I have hardware echo cancelling) and it didn't do anything. I'm forcing his codec to G729 for bandwidth reasons. The phone is an Aastra 6757iCT. Does anybody have any suggestions here? -- _ -- 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] Clipping issue with SIP over satellite
You have hardware echo canceling *outside* of your T1 card? No, on the card. The DAHDI layer has some buffering that can help with jitter, but the default buffers can only handle 80ms of jitter. You can increase this by setting the 'buffers' option in chan_dahdi.conf; each buffer is 20ms by default. I'm running 1.6.2 and it appears that this is called jitterbuffers there. Is that right? I've set it to 20 and it did indeed help quite a bit, so I tried 30. It sounds like the lack of a proper jitter buffer (of adequate size) is the issue here, since when the audio is directed at endpoints outside of Asterisk that have them, the audio is as you'd expect it to be. Interestingly, that isn't completely true. If it goes out a SIP trunk to PSTN, it works fine, but when it goes out a SIP trunk to the SV8300 (where the T1 goes), it has the same problem. This was leading me to believe that the problem was on the 8300. -- _ -- 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] Clipping issue with SIP over satellite
You have hardware echo canceling *outside* of your T1 card? No, on the card. Then you definitely don't want 'echocancel=no' set, or you'll disable it. When I thought that it was echo cancellers fighting each other, that's exactly what I wanted to do. -- _ -- 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] One-way audio with media_address
I'm migrating from Asterisk 1.6.2 to 10.7.0. In 1.6.2, I made a small patch to allow specifying an address for RTP media. That worked. In 10.7.0, this appears to be built in with media_address, but it doesn't work for me. My Asterisk server has multiple addresses, all global address on two different /24's with different routing policies via BGP. I'm connecting to a phone that's over NAT. I have nat=yes in the general section of sip.conf. Everything works fine with the default. But if I specify media_address to be the Asterisk server's address on the other /24, I get one-way audio. I can see with sip debug that the proper address is being given in the SDP data. Audio from the phone is fine. Audio *to* the phone starts out with maybe 1-2 seconds of very garbled audio, then goes quiet. Running traceroute shows that data comes from the phone *to* Asterisk on the desired /24, but goes out with a source address from the other /24 (the default address). I'm not sure if this is the problem or not, but in any event, I think the source address for RTP should be the one in media_address and want it that way for my purposes anyway. Is there a way to configure this to happen? If not, where should I look to make a patch? And is this likely the reason for the one-way audio or is something else the likely cause? -- _ -- 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] Repeated Asterisk 10.7.0 crashes
I'm getting cycles of repeated crashes which occur and then stop occurring. Looking at the dumps via gdb shows that something peculiar is happening that looks like memory corruption: Program terminated with signal 6, Aborted. #0 0x003686e30285 in raise () from /lib64/libc.so.6 (gdb) up #1 0x003686e31d30 in abort () from /lib64/libc.so.6 (gdb) up #2 0x003686e6971b in __libc_message () from /lib64/libc.so.6 (gdb) up #3 0x003686e71e7e in _int_malloc () from /lib64/libc.so.6 (gdb) up #4 0x003686e7382d in calloc () from /lib64/libc.so.6 (gdb) up #5 0x0054a2a0 in _ast_calloc (num_structs=1, struct_size=88, field_mgr_offset=64, field_mgr_pool_offset=16, pool_size=128, file=0x101010101010101 Address 0x101010101010101 out of bounds, lineno=1235, func=0x58af9e ast_log) at /usr/src/asterisk-10.7.1/include/asterisk/utils.h:495 495 AST_INLINE_API( Once this starts happening, it seems to keep happening, but Asterisk seems to stay up for hours between the cycles, which I can't reliably stop from cycling. Does anybody have any ideas how to debug this? I suspect it may have something to do with either res_speech_lumenvox (which I got from Lumenvox) or res_speech_unimrcp (which looks to be extremely buggy). -- _ -- 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] Responsibility for res_speech_lumenvox.so
Who's responsible for it? Lumenvox is the only place that distributes it, but they can't do anything with it since they get it from Digium. However, the current version doesn't work with Asterisk 10.7.1 and the latest version of Lumenvox software (it appears that a timeout is being set to zero). -- _ -- 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] Any workaround for res_speech_lumenvox.so issue?
The latest version of res_speech_lumenvox.so doesn't seem to work and nobody seems to know when a version that works will be available. It looks to me like this is some sort of timeout issue. Does anybody have a workaround to allow this to be used? (I know about UniMRCP, but find it quite heavy.) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Questions on converting to ConfBridge
I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so it becomes very hard to have web applications that show who's in a conference. There also doesn't seem to be a way to lock conferences or mute or kick out users from the dialplan. And the CLI command needs a channel, not a user index, making scripting via the dialplan that much harder. What am I missing? -- _ -- 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] 10.9.0-rc1 : Help with GoSubIf Parsing
I'm getting a parsing error with the folllowing:
same=n,GoSubIf($[${CALLERID(num)} = 2024324321]?other,1($
{thisexten}):)
WARNING[11356]: ast_expr2.fl:468 ast_yyerror: ast_yyerror(): syntax
error: syntax error, unexpected '=', expecting $end; Input:
= 2024324321
I've tried with and without spaces the = sign. Same Result. I've
counted my parens and braces.
If there *is* a caller-ID, it should work without spaces. But not if
there isn't. The proper test is:
$[x${CALLERID(num)}=x2024324321]
And this only works if you're *sure* that it'll be just numbers or blank.
Otherwise, use quotes on both sides.
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[asterisk-users] Question on Asterisk memory management
I'm trying to add a Talking: field to the AMI ConfbridgeList event so that my conference room monitoring will work with Confbridge instead of having to stay with MeetMe and there's something I don't understand. When app_confbridge.c calls ast_bridge_features_set_talk_detector, it passes a *copy* of args.conf_name. Why make the copy? Isn't args.conf_name in valid memory throughout the existance of that bridge? I ask because the easiest way to do what I want is to change that parameter to be conference_bridge_user and add a talking field to it (yes, I know I then have to have the callback called unconditionally and test TALKER_DTETECT there). But that can't work if there's a scoping issue with memory and the copy suggests there is, though I don't see it. -- _ -- 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] Odd cracking with SIP-DAHDI
We recently set up a SIP trunk between an office in NY running Asterisk and an office in Paris (running Alcatel). All works fine if a SIP phone on the NY system talks to the Paris PBX. But if something on DAHDI (a PRI or MeetMe) talks to the Paris PBX, there's a low-volume crackling. This isn't clipping because it also occurs when there's no legitimate sound. It's sort of a mild version of what you used to get when a POTS pair had a ground short. This occurs no matter what size originates the call. pings show round trip times of around 100ms, ranging from around 200 to 80 ms. Packet loss is zero. The fact that SIP-SIP works fine suggests the issue isn't related to IP issues. I tried adding a jitter buffer, but that didn't make a difference. I've tried this sending just ULAW and G722 and allowing everything, but no difference. The SDP that comes back from Paris doesn't list any audio codecs and is: v=0 o=default 1350406175 1350406175 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 0 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 m=video 0 RTP/AVP 31 34 34 98 99 104 a=rtpmap:31 H261/9 a=rtpmap:34 H263/9 a=rtpmap:34 H263/9 a=rtpmap:98 h263-1998/9 a=rtpmap:99 H264/9 a=rtpmap:104 MP4V-ES/9 a=sendrecv Does anybody have any ideas as to what I should look at next? -- _ -- 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] Odd cracking with SIP-DAHDI
cat proc/interrupts? http://wiki.openvox.cn/index.php/Troubleshooting_of_PRI_cards I'm sorry that I wasn't clear: the PRI is fine. It's been in use for years and hasn't caused any problems. What's new is the SIP connection between the two offices. And another datapoint: the problem only happens for ulaw and alaw, not g729. -- _ -- 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] Odd cracking with SIP-DAHDI
I seem to recall seeing somewhere recently where there was a bugfix for ulaw/alaw conversion which would cause poor audio. Hmm. You mean: https://issues.asterisk.org/jira/browse/ASTERISK-1323 That was quite old, but that is what the noise sounds like. Have you tried updating your Asterisk to the latest of whatever major version you are running? I'm running 10.7.1, which is pretty new. I'd prefer not to upgrade unless I know it'll fix it because of the work involved. -- _ -- 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] Wierd RTP issue
I have a peculiar RTP issue. I'm experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it's connected via NAT, all is OK. When it's connected with VPN, the following occurs: The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it's calling. The voice path inbound to Jitsi works fine when it's called from another SIP device. The voice path inbound to Jitsi is silent when it's called from something on the other side of a PRI via DAHDI. I've run Wireshark on my desktop and see the RTP packets coming at the same rate and protocol (g711) in all the cases and sip set debug peer xyz doesn't shed any light on the situation (the SDP data looks similar in the working and non-worknig cases). Does anybody have any ideas what to look at next? -- _ -- 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] Wierd RTP issue
What's the configuration like for Jitsi in sip.conf? Just fullname and md5secret plus a phones section that reads: [phones](!) type=friend host=dynamic context=SIP_Phones cc_agent_policy=generic cc_monitor_policy=generic disallow=all allow=gsm allow=ulaw allow=g729 allow=h264 What version of Asterisk? 10.7.1 What does the SIP signaling look like? I don't follow. It's just the standard INVITE/Ring/OK. -- _ -- 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] Wierd RTP issue
What NAT settings are globally in use? nat=yes Do you have directmedia turned off or on? I've tried both ways, but I normally have it off. This really does indeed feel like a weird NAT issue that is probably configuration related (probably both in Jitsi and Asterisk). Except that: (1) It *works* when there's NAT and *doesn't* work when everything has a static IP. (2) I see the RTP packets arriving: if it were NAT, I'd expect *not* to see them. (3) It depends on the direction of the call and on whether it's SIP-SIP or DAHDI-SIP (and directmedia is off). -- _ -- 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] Wierd RTP issue
Yeah this is so weird that packet captures are really needed. A working call and a non-working call, along with what IP ranges are what. There are *tremendous* numbers of RTP packets, of course. Are those captures really going to be useful? That's the problem. If they *are* going to be useful, then how many packets should I save? I did look at the sip debug output, as I said, and those look the same. I ran into this on a machine that I won't be at for another two weeks, but I can see if I can reproduce it on similar machine. -- _ -- 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] Wierd RTP issue
Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I'll try to reproduce on this machine and send that off. However, I did look at the SIP signaling and src/dst IP addresses and they're all as expected between the two calls: I really fear that the difference is in the contents of the RTP stream. -- _ -- 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] Wierd RTP issue
Not that many RTP packets are required. It's just important to see the SIP signaling and where traffic is coming/going from with the network topology in mind. That way a clearer picture of where it's saying media should go to, where it's sending media from, etc can be gleamed. Once that is figured out then the problem can be isolated. OK, I reproduced it on this machine. It's a total of only 1293 packets, taken on this end. First call didn't work: I heard nothing coming inbound. Second call worked, well enough that there was feedback (both phones and the desktop were in the same room). You can find the file at: http://www.gnat.com/~kenner/wierdAsteriskJitsi.pcap -- _ -- 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] Wierd RTP issue
1. Remove allow=gsm from your sip.conf and reload That did it! Thanks! But why should that have been an issue? 2. Disable ZRTP in Jitsi by going into Options - Accounts - Selecting account - Edit - Security - Uncheck Enable support to encrypt calls. That was one of the first things I tried a few days ago. No change. -- _ -- 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] Wierd RTP issue
The way you had things configured Asterisk was prioritizing GSM over ULAW, so until Jitsi started responding it sent GSM. I thought I might have seen something like that in the packets, but it didn't look like it showed up in the SDP negotiations, so seemed peculiar to me. Unclear why this only happens with a static IP and not NAT, but 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
Re: [asterisk-users] Top Posting
I realize the benefits of bottom-posting, especially when posting inline. But top-posting keeps things in reverse chronological order so any reader could catch up quickly on any missed messages in the chain. A new reader scrolls to the bottom and reads up. What's there to catch up with if you don't first read what the person is replying to? Do you think that everybody remembers every thread. Of what value is it to see something like No, that didn't work. *before* a description of what it was that didn't work. When people reply to an email, it's their responsibility, whether they top-post or bottom-post to remove unnecessary old message and keep just what's necessary to understand the email. One of the problems with top-posting is that it makes it easier to forget to do this. -- _ -- 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] Problem with Speex codec
I'm trying to convert from MeetMe to Confbridge and one part of that is
handling the ending of a conference. So I'm taking the suggestion of
originating a call to the conference and doing:
same = n,Playback(conf-will-end-indigits/${WTIME}minutes)
That crashes Asterisk (with no core dump!) in the default configuration.
When I run it manually, I see the error message:
Fatal (internal) error in kiss_fft.c, line 294: KissFFT: max radix supported is
17
If I unload module codec_speex.so, everything works. If I playback files
other than conf-will-end, it also works.
Two questions:
(1) Why is that codec being used in the first place?
(2) Why it is generating that error when it is?
This the Asterisk 10.7.1 release.
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[asterisk-users] Question on Confbridge menu item dialplan_exec
I like the example of using that to add somebody to the conference, but what I don't see is how the dialplan can know what conference the menu item was called from. I was hoping that some variable might have been set, but don't see it in the sources. Is the idea to do that outside of the call to Confbridge? -- _ -- 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] Top Posting
I'm the opposite. I'm likely not to scroll down 10 pages to see the comments at the end. Wouldn't need to if people trimmed their posts properly. Precisely (e.g., see above)! Indeed, my sense is that top-posting *discourages* properly trimming email and that's my main reason against it. If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. -- _ -- 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] Top Posting
In this properly trimmed example, there's no record of who said what. When it's relevant, I trim in such a way that that information is preserved. But I would *never* leave in a header, just the identification of the person who typed that part. Most mailers, when you include text from another email, put someting like XYZ wrote: before the included text. So usually it's just a matter of preservating that and adding any that are needed that aren't there. Yes, it takes a few minutes longer, but given that there are probably hundreds of people reading my email, that's an investment that I find *well* worth it. -- _ -- 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] Top Posting
If things were properly trimmed, the email would be short enough that it really doesn't matter that much if the new material is on the top or bottom, but people who top-post and don't trim create really hard-to-follow emails. Not really true often times when people do the right thing and post debug and conf files often required to get meaningful help. Yes, but if you put those at the end, where they belong, people reading the email can follow the thread quite easily and can ignore those if they don't need them. Certainly only a tiny part of such, if any at all, should be included in a reply. -- _ -- 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] Any timeframe for the release of the Asterisk 11-Lumenvox connector bridge?
I'm starting to think about migrating from an old Asterisk box to a new one and want to use the Asterisk 11 long term support release, but need Lumenvox integration and I don't see the Asterisk 11 connector bridge for Lumenvox available yet. Lumenvox tech support says this is under Digiums control. Can anyone give an idea of how soon it'll be available? I will need this as 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] Problems with 'i' extension
I'm running Asterisk 10.7.1. In the log, I see:
-- Goto (Conferences,70323,1)
-- Auto fallthrough,
But there is an 'i' extension:
dialplan show i@Conferences
[ Context 'Conferences' created by 'pbx_config' ]
'_[ti]' =1. GotoIf($[${SET(REC=$[${REC}--1])}3]?999) [pbx_config]
2. Set(EFN=conf-invalid) [pbx_config]
3. Goto(200,1)[pbx_config]
What's going on? Shouldn't this go to that extension?
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[asterisk-users] Uninitialized variable in main/pbx.c?
I think the below fixes what I reported earlier. Does that seem right? *** pbx.c.old 2013-01-23 21:08:51.0 -0500 --- pbx.c 2013-01-23 21:09:31.0 -0500 *** static enum ast_pbx_result __ast_pbx_run *** 5160,5163 --- 5160,5165 int timeout = 0; + dst_exten[0] = '\0'; + /* loop on priorities in this context/exten */ while (!(res = ast_spawn_extension(c, c-context, c-exten, c-p riority, -- _ -- 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] Uninitialized variable in main/pbx.c?
+ dst_exten[0] = '\0'; Is this 'construct' prefered over dst_exten[0] = 0; or *dst_exten = 0; and why? I'm somewhat of a C pedant here. dst_exten is declared as an array, not a pointer. So if I want to clear the first byte of the array, I'll use array syntax pretty consistently. If it's a pointer, I tend to prefer the pointer syntax, unless I'm also doing something with other than the first byte. So I wouldn't write: *x = 'a'; x[1] = '\0'; but instead x[0] = 'a'; x[1] = '\0'; And I certainly don't like using 0 when I mean the null character, at least not in an assignment. -- _ -- 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] g723 transcoding
It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that? -- _ -- 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] clicking sound with alaw codec
I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. The outgoing SDP looks like this: v=0 o=root 1691755711 1691755711 IN IP4 205.232.38.178 s=Asterisk PBX 10.7.1 c=IN IP4 205.232.38.178 t=0 0 m=audio 11432 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv The reply SDP is: v=0 o=default 1359060187 1359060187 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 Any suggestions on how to debug what's causing this? -- _ -- 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] clicking sound with alaw codec
Your sounds might be too loud. We use a lot of custom sounds here and when the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and clicks. Sorry I wasn't clear. This is *always*. I hear it over the call when there's talking and when there's dead silence (e.g., an empty MeetMe room). -- _ -- 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] clicking sound with alaw codec
When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come to mind in no particular order are: Remember: this is only *one* particular SIP trunk. Use Wireshark to see the difference between a good call and a bad one. If you see a lot of time jumps on the bad call then look at your network/QoS. jumps? Note that a good call is G.729 and bad is G.711, so I wouldn't expect them to be at all similar. We throw a lot more bandwidth than even G.711 down the pipe between the two sites in terms of data each evening, so I don't think it's that kind of issue. I'm thinking in terms of distortion caused by transcoding someplace. -- _ -- 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] clicking sound with alaw codec
- jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371424, src=RTP [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371434, src=RTP -- _ -- 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] clicking sound with alaw codec
Check https://issues.asterisk.org/jira/browse/ASTERISK-12042 I did. But that was with an unofficial G.729. This is with the supplied alaw codec. -- _ -- 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] Frames with invalid timing info
I'm now getting these errors: [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891164, src=RTP [Jan 25 09:19:01] WARNING[29877]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-ba7 received frame with invalid timing info: has_timing_info=1, len=0, ts=426891174, src=RTP even *without* any transcoding. Suggestions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with .siren14 sound files
I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14. I downloaded the codecs and now it will properly transcode to connect to other phones and play any files that are in .wav format. But when it tries to play any files with .siren14 extensions, I get complete noise coming out. Here's the negotiated SDP: v=0 o=root 1668560220 1668560220 IN IP4 207.10.184.50 s=Asterisk PBX 10.7.1 c=IN IP4 207.10.184.50 t=0 0 m=audio 16204 RTP/AVP 115 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:127 telephone-event/8000 a=fmtp:127 0-16 a=ptime:20 a=sendrecv If I rename away the .siren14 files, all is OK. I can't find anything related to this with a search. -- _ -- 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] Transcoding issues with siren14
Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and slin16 to slin32. So why is slin used as the intermediate instead of slin16? -- _ -- 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] Transcoding issues with siren14
Do you have transcode_via_sln set in asterisk.conf? No, but as I said in a later email, I found the problem: when computing the cost of a path, any downconvert has the same cost. So siren14 - slin - slin32 is the same cost as siren14 - slin16 - slin32 which is wrong. I fixed this by adding the magnitude of the difference in the sampling rate to the cost, but I'm not sure if that's the right solution. -- _ -- 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] Disagreements between codec_siren14 and Polycom sources
There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. If I make a .sln32 file and run the encoder from ITU/Polycom with encode 0 foo.sln32 foo.siren14 48000 14000 the resulting file doesn't play back correctly with the Digium's siren14 codec. I know the parameters are correct because the file is the same size as that made by the Digium codec. Both sets of decoders/encoders (Digium and Polycom/ITU) are symmetric and can decode what they encode, but neither can read the encoding of the other. Is there some subtle difference between G.722.1C and Siren14? -- _ -- 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] Disagreements between codec_siren14 and Polycom sources
I'm answering my own email here: There appears to be a disagreement between the encoding given in the sources for Siren14 that are downloaded from Polycom (and the ITU, both are the same) and that implemented by codec_siren14.so. The latter agrees with the actual device. The disagreement is in byte-swapping of the encoded stream. Once that's done, things work fine. If anybody wants a codec that can transcode between Siren14 and slin32 (which is better than Digium's codec_siren14 codec which goes to slin and slin16), let me know. I can send a file that calls the Polycom/ITU code. -- _ -- 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] Integration with skype
For voice, you can use SipToSis. Works flawlessly with Asterisk and the best part, it's free. :) www.mhspot.com/sts/ (site is down right now) And that's related to the problem with it: it hasn't been maintained for quite a while. -- _ -- 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] Jitter buffer on write side of channel
How does one do this? We have a particular SIP phone that needs a large jitterbuffer, but all I can see is how to put it on the *read* side of the channel. -- _ -- 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] Recording conferences with changing bitrate
I'm running 10.7.1 (yes, I know it's old, but this may be a problem in later versions too) and had a conference being recorded via: Set(CONFBRIDGE(bridge,record_conference)=yes) The bridge started out at 8KHz despite one HD device. But when the second came in (G.722), it switched to 16KHz. At that point, the recording file had the bitrate change in the middle. That seems wrong. I'd expect the bitrate of the recording channel to remain unchanged and transcoding to be used to do the recording. But it wasn't. Does this ring a bell with anybody? -- _ -- 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 CLI Banner
Modifying a program you have legitimately acquired is Fair Dealing. The Law of the Land gives you the right to do that, even if the vendor restricts your exercise of that right in practice by withholding the Source Code. That is false. Modifying a program is creating a derivative work. As purchaser of a copyrighted item, you normally *do not* have that right. And this certainly may vary from jurisdiction to jurisdiction. For a (quite dated at this point) discussion of this issue from a US perspective, see http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157 The author is a recognized expert in software IP law. -- _ -- 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 CLI Banner
What does violating license of Asterisk means? Does it means I won't be able to use any commercial modules or asterisk commercially? I thought it was open and anyone can change the code? Anyone *can* change the code. But it's licensed software, just like most other software. The difference is that the GPL gives you rights that you don't have for other non-open software. However, in both cases, you have to be sure that you don't violate the terms of the license. If you're unclear as to whether what you propose to do will violate the license, I'd suggest consulting an attorney: nobody on this list (or any other) should be providing you legal advice. -- _ -- 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 CLI Banner
Of course, any good attorney will never commit to anything. They will never say it is alright to do X, unless X is do nothing No, but a good attorney can give guidance as to likely expectations. As you say, nobody can be sure of something even if it's previously been established law, but a good attorney can point out potential pitfalls on the one hand and identify, on the other, things that are much less likely to be an issue. It's not a guarantee, but you can often get a recommendation about whether or not it's a good idea (not necessarily alright) to do something. Attorneys often have to a take a stand on these matters. If a company needs to use software that performs a specific thing and, say, only three companies provide such, but under different licensing terms, it's the job of that company's legal department to say which, if any, they can be used. Doing nothing will have a cost and risk here too because this example is talking about something that the company needs done. -- _ -- 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 CLI Banner
If you really want to do it: 1) create a wrapper to asterisk -r 2) pipe the welcome message to /dev/null 3) ??? 4) profit you didn't modify Asterisk. No you didn't, but you may neverthess have created a derived work. There are two different legal arguments you can make when two pieces of software are tightly coupled in that way: one argues that it's a derived work and the other that it's not. Copyright law when it comes to software is not simple and certainly not obvious. If you want to use a piece of Free Software in a commercial product, you need to consult an attorney. It's really that simple. -- _ -- 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] Problem building Asterisk-12.2.0
When I run ./configure, it aborts with: checking for uuid_generate_random in -luuid... no checking for uuid_generate_random in -le2fs-uuid... no checking for uuid_generate_random... no configure: error: *** uuid support not found (this typically means the uuid development package is missing) But it *is* installed: [root@asterisk asterisk-12.2.0]# yum list installed | grep uuid uuid.i386 1.5.1-3.el5 installed uuid.x86_64 1.5.1-3.el5 installed uuid-devel.i386 1.5.1-3.el5 installed uuid-devel.x86_64 1.5.1-3.el5 installed So I'm confused ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk-12.2.0
I think you need the libuuid and libuuid-devel packages. yum list available was not showing any such package. I installed a few other packages, including uuid-dce-devel and one of them did the trick, but the install-prereq script wasn't good enough. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk-12.2.0
What distro are you building on? CentOS 5.10. Both have the libraries listed in install_prereq. Indeed it has all but 2 or 3 of those libraries (none related to uuid), but after running that script, it was still missing what it needed for uuid. Unfortunately, there's no upgrade path from CentOS 5.10 to 6.5. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem building Asterisk-12.2.0
e2fsprogs-devel is the package that provides uuid.h on centos 5 I tried that first and it didn't seem to. I'm pretty sure I needed uuid-dce-devel. -- _ -- 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] CBAnn channel not going away in Asterisk 12
After an upgrade to Asterisk 12, I'm collecting channels. When I enter and then exit a conference room, I see: -- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language 'en') -- Channel CBAnn/207-067f;2 joined 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c -- Channel CBAnn/207-067f;2 left 'softmix' base-bridge 5edb1920-3774-4ba3-8c4d-23e8fd04519c I'd expect those channel to immediately go away, but they just stay around: asterisk*CLI core show channel CBAnn/207-067f;1 -- General -- Name: CBAnn/207-067f;1 Type: CBAnn UniqueID: 1398809161.20186 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (nothing) WriteFormat: unknown ReadFormat: unknown WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h1m3s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:07:04] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;1 asterisk*CLI core show channel CBAnn/207-067f;2 -- General -- Name: CBAnn/207-067f;2 Type: CBAnn UniqueID: 1398809161.20187 LinkedID: 1398809161.20186 Caller ID: (N/A) Caller ID Name: (N/A) Connected Line ID: (N/A) Connected Line ID Name: (N/A) Eff. Connected Line ID: (N/A) Eff. Connected Line ID Name: (N/A) DNID Digits: (N/A) Language: en State: Up (6) NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Time to Hangup: 0 Elapsed Time: 0h3m30s Bridge ID: (Not bridged) -- PBX -- Context: default Extension: s Priority: 1 Call Group: 0 Pickup Group: 0 Application: (N/A) Data: (Empty) Call Identifer: (None) Variables: [Apr 29 18:09:31] ERROR[21102]: cdr.c:3106 ast_cdr_serialize_variables: Unable to find CDR for channel CBAnn/207-067f;2 -- _ -- 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] CBAnn channel not going away in Asterisk 12
The announcer channel joins/leaves the conference as it has sounds to play. If the channel still hangs around after the conference is destroyed then there is a problem. There's a problem. ;-) But thanks for pointing to how that's supposed to be handled. -- _ -- 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] CBAnn channel not going away in Asterisk 12
If the channel still hangs around after the conference is destroyed then there is a problem. Am I missing something obvious: I'm looking in the confbridge_exec function. I see a conference = NULL line, but no attempt to free that structure, which is what I understand will destroy the playback channel. So where it is freed? -- _ -- 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] CBAnn channel not going away in Asterisk 12
If the reference count on the bridge is off, you should see the conference bridge 'hanging around' after the last participant has left. And how would I be sure this is the case? I did core set debug 1 and didn't see the debug line about destroying the conference, but it doesn't show up in confbridge list. -- _ -- 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] CBAnn channel not going away in Asterisk 12
Really, I think we're pretty positive there's a ref leak (since otherwise, the CBAnn channel would be long gone). If you can get a ref debug log and the standard Asterisk DEBUG log showing the problem, that would help a lot in finding out what is going on. That can't be done in the 12.2.0 release, just the current SVN, right? Clearly this occurs for me and not in the simple case. So I think what I'll do is see exactly what I have that's causing it and hopefully code inspection of that piece will show the missing ref decrement. I'm away for a few days and so may not be able to get to this until I get back. Thanks for the pointers. -- _ -- 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] CBAnn channel not going away in Asterisk 12
It may show up in 'bridge show all' - but I'd actually expect it not to show up there either. Actually, it does. I have a screen full of bridges with 0 channels. I just tried an experiment where all I have is exten = 329,1,Answer(1000) same = n,Confbridge(1234) with absolutely nothing else going on and those leak too. I need to understand why I'm seeing this and nobody else is. -- _ -- 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] CBAnn channel not going away in Asterisk 12
Please go ahead and open an issue and attach the refs log and the full DEBUG log. That will allow us to understand what's occurring here. I need to wait until I'm sure this isn't something I caused somehow, so I need to first understand why I'm seeing this and nobody else is. -- _ -- 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] CBAnn channel not going away in Asterisk 12
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a lot in finding out what is going on.
I think the bug is in conf_handle_talker_cb. It calls ao2_find but has no
mechanism to decremement the refcount. It appears that the following is
the best fix. I looked at all remaining calls to ao2_find in app_confbridge.c
and they look OK. Do you agree with the below fix?
*** app_confbridge.c.bug2014-05-06 06:42:21.0 -0400
--- app_confbridge.c2014-05-06 06:42:05.0 -0400
*** static int conf_handle_talker_cb(struct
*** 1461,1467
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! struct confbridge_conference *conference = ao2_find(conference_bridges,
conf_name, OBJ_KEY);
struct ast_json *talking_extras;
if (!conference) {
/* Remove the hook since the conference does not exist. */
--- 1461,1468
struct pvt_talker_cb *pvt = hook_pvt;
const char *conf_name = pvt-conf_name;
! RAII_VAR(struct confbridge_conference *, conference, NULL, ao2_cleanup);
struct ast_json *talking_extras;
+ conference = ao2_find(conference_bridges, conf_name, OBJ_KEY);
if (!conference) {
/* Remove the hook since the conference does not exist. */
--
_
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Re: [asterisk-users] CBAnn channel not going away in Asterisk 12
That is definitely a leak and the fix looks good. Thanks. That leak is most likely the one biting you. It definitely is. There is another leak in handle_cli_confbridge_kick() if the participant to kick is not in the conference. Confirmed. I missed that one in my code reading. I just fixed it the same way. Please go ahead and open an issue so proper credit can be given for the patch. I'm not concerned about credit, but would like to get it fixed. I need to figure out what has to happen for me to be able to submit patches, but then I'll have some others to submit too. -- _ -- 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] CBAnn channel not going away in Asterisk 12
Committed the fix for this leak on Asterisk v12 branch in -r413452. This leak also applied to Asterisk v11. Thanks. Is this for both the one in the talking callback or the one in handle_cli_confbridge_kick or both (the fix is similar in both)? -- _ -- 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] WSS over Asterisk
I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event = failed_to_start Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works fine. Any idea why? Sorry for the delay in answering: I meant to reply and forgot. ws:// uses HTTP and wss:// uses HTTPS so there's no way they can work via the same socket. You have to set up a separate HTTPS socket for wss. -- _ -- 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] Getting source ip adress of incoming INVITE
I'm interested in finding out what the source ip is of an invite in the
dialplan (Asterisk 11).
${CHANNEL(recvip)}
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Re: [asterisk-users] SIP Jitterbuffer
What are the cons, if any, of enabling a jitterbuffer? Memory and latency. -- _ -- 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] default features
Question: is there some built-in way to know if macro feature1-ClientA is defined? Something liken ExecIfMacro(feature1-ClientA)?macro(feature1-ClientA):Goto(...). A macro is a context, so DIALPLAN_EXISTS should work if you specify an extension and priority that's in the macro (presumably, s,1). -- _ -- 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] setting outbound caller ID
CALLERID is a read only variable. That's not correct. I set it all over the place in my dialplan. -- _ -- 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] Bug in ast_frame_adjust_volume in 12.2.0?
I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line:
351 res = (int) *input * *value;
It's called from ast_frame_adjust_volume.
The frame looks like:
(gdb) print *f
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = {
id = AST_FORMAT_SLINEAR16, fattr = {format_attr = {
0 repeats 64 times}, rtp_marker_bit = 0 '\000'}}}, datalen = 0,
samples = 320, mallocd = 1, mallocd_hdr_len = 1076, offset = 64,
src = 0x51623b0 func_jitterbuffer interpolation, data = {ptr = 0x0,
uint32 = 0, pad = \000\000\000\000\000\000\000}, delivery = {
tv_sec = 1436290187, tv_usec = 304285}, frame_list = {next = 0x0},
flags = 0, ts = 0, len = 0, seqno = 0}
so datalen is 0 and samples nonzero. ast_frame_adjust_volume, however,
iterates over samples, not datalen. Is that correct?
What does it mean to have a packet with a zero datalen anyway?
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[asterisk-users] Siren7 and Asterisk 13
I'm planning on upgrading to Asterisk 13.4 soon and am looking for the corresponding Siren7 codec. Where do I find it? -- _ -- 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] Siren7 for Asterisk 13.5
What is the proper version of the Siren7 codec to use for Asterisk 13.5.0? Since there's nothing later, does the version for 12.0 work? -- _ -- 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] Siren7 for Asterisk 13.5
A Siren codec is not currently available and the one for 12 will not work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why). -- _ -- 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
