Re: [asterisk-users] Removing mailbox and password prompt for voicemail
On Sat, Jul 30, 2016 at 7:18 AM, D'Arcy J.M. Cain wrote: > > Bad, bad idea. If you remove the password then anyone can get to the > mailbox. Depends on your use case, at home I have several phones and one mailbox. So I _want_ everyone to get to the mailbox with a minimum of effort. Brian -- _ -- 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] Removing mailbox and password prompt for voicemail
On Fri, Jul 29, 2016 at 10:43 PM, Nabeel wrote: > Hello, > > I am using Asterisk voicemail on a CentOS 7 server. I would like to be > able to remove the 'mailbox' prompt and 'password' prompt when a user tries > to access their voicemail. I can remove the 'password' prompt by not > setting a password for the user, but the 'mailbox' prompt is always heard. > Please let me know how Asterisk can be configured to remove these prompts. > > Nabeel > It sounds like you really want it to jump straight into one voicemail box with no auth, not just removing the prompts. Try something like this exten => *85,1,VoicemailMain(100,s) When you dial *85, you will get voicemail for mailbox 100, and no passcode prompt. Jumps straight into the menu "You have FOUR new messages..." -- Brian Wilson Wildsong -- _ -- 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] Remove 'Comedian Mail' Message
On Fri, Jul 29, 2016 at 11:01 PM, Nabeel wrote: > Hello, > > I would like to remove the 'Comedian Mail' name/message played when a user > tries to access their voicemail. Please let me know how to do this. > > Nabeel > It plays a file /var/lib/asterisk/sounds/en/vm-login.wav The recording says "Comedian Mail, please log in now." We thought we still needed a voice prompt, so we recorded a new message. I am not sure what happens if the file is missing, try removing it and test. Alternatively the file could be empty (0 seconds of audio) or have a tone or some other sound in it as feedback. The file has to be in the right format for asterisk, after recording it on a Windows machine, I used this command to convert it: sox infile.wav -r 8000 -e signed-integer -b 16 -c 1 outfile.wav "Sox" figures out the input format. -- Brian Wilson Wildsong -- _ -- 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] __sip_xmit Returned -1 Invalid Argument
Reviving an old thread, still seeing this. Brian Wilson wrote: >* I've been getting slammed with these messages on my console lately. *>>* ed -1: Invalid argument *>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit: *>* sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0 <http://192.168.1.45:0> *>* <http://192.168.1.45:0 <http://192.168.1.45:0>> returned -1: Invalid argument *>* [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit: *>* sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0 <http://192.168.1.45:0> *>* <http://192.168.1.45:0 <http://192.168.1.45:0>> returned -1: Invalid argument * Joshua Colp responded: > It's an invalid argument because the port appears to be 0, > which won't work. How it got to be that I don't know... a full log may provide further enlightenment. Currently I have 9000+ messages similar to this: [2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 725) to 192.168.89.172:54490 returned -2: Success [2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 725) to 192.168.89.172:54490 returned -2: Success [2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 727) to 192.168.89.172:54490 returned -2: Success [2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 728) to 192.168.89.172:54490 returned -2: Success [2016-07-25 23:06:52] WARNING[1993] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 729) to 192.168.89.172:54490 returned -2: Success [2016-07-25 23:06:53] WARNING[1993] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 729) to 192.168.89.172:54490 returned -2: Success [2016-07-25 23:07:04] WARNING[14771] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 711) to 192.168.89.172:54499 returned -2: Success [2016-07-25 23:22:33] WARNING[15801] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 711) to 192.168.89.172:54502 returned -2: Success [2016-07-25 23:22:33] WARNING[15802] chan_sip.c: sip_xmit of 0x7f1ce8687450 (len 711) to 192.168.89.172:54502 returned -2: Success Port is not zero on these. Error code has changed from -1 to -2 What do you mean by a "full log". Thanks -- Brian -- _ -- 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] 1 way audio but audio+video is fine
Sporadically we get 1 way audio when one party is outside our firewall. The caller is on NAT, and it works fine most of the time. Caller can hear the called party, same thing going the other direction. Caller can hear called party. Asterisk 13.9 on Debian chan_sip with two identical Grandstream GXV3240 SIP phones In the CLI, I can see calls on our PBX running media on port 5004 when the call is audio only. When the caller switches to video calls it opens ports 5004 and 5006 and everything works fine. This does not make sense to me. It seems like audio on port 5004 should fail either way -- adding another media channel for video should not affect audio. I believe I have UDP ports 5000-4 open right now on the firewall. I also don't understand why it varies from day to day. Any ideas on how to debug or what might be happening? -- Brian Wilson, GISP Wildsong: 707-827-0001 Mobile: 707-332-3521 -- _ -- 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] SPA112 flapping
I gave up on a couple devices here in my home network (an old SPA2000 and a cheap DP715 grandstream) and plugged their IP addreses into Asterisk and turned off registration. No more annoying messages. On Fri, Jun 17, 2016 at 2:56 PM, Mike Diehl wrote: > Hi all, > > I've got a device that seems to become unreachable for about 2 minutes, > every > hour. From what I can tell, it isn't due to network or server issues. Any > ideas? > > TIA. > > > -- > Mike Diehl > Diehlnet Communications, LLC. > Voice: (505) 903-5700 > Fax: (505) 903-5701 > > > -- > _ > -- 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 > -- Brian Wilson, GISP Wildsong 707-827-0001 -- _ -- 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] __sip_xmit Returned -1 Invalid Argument
I've been getting slammed with these messages on my console lately. ed -1: Invalid argument [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit: sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0 returned -1: Invalid argument [2016-05-31 10:09:40] WARNING[16249]: chan_sip.c:3775 __sip_xmit: sip_xmit of 0x7f05140803f0 (len 559) to 192.168.1.45:0 returned -1: Invalid argument [ etc etc etc comes in bunches of about 10, all with same IP address in that bunch. Does not appear to be related to one phone -- I have a mix of devices. They appear to happen when a phone registers. The debug message "Invalid argument" is not helping me, I wish it were more descriptive. I peeked at the code and there is no indication what __sip_xmit() does (like all Asterisk code I have looked at so far), looks like it might do MANY things. It does not appear to be tied to one version, I have tried 13.7.2, 13.8.0, 13.9.1 I don't see anything glaring when I "sip set debug on" and "core set debug 10" -- just more messages that I don't understand. Like this -- --- [2016-05-31 10:10:39] WARNING[16249]: chan_sip.c:3775 __sip_xmit: sip_xmit of 0x7f0514050430 (len 559) to 192.168.1.45:0 returned -1: Invalid argument Retransmitting #5 (no NAT) to 192.168.1.45:0: NOTIFY sip:spa2000@192.168.1.45:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK0d196419 Max-Forwards: 70 From: "asterisk" ;tag=as522feace To: Contact: Call-ID: 0d6963aa7350ce3274e7af420c35daa6@192.168.1.2:5060 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 13.9.1 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 91 Messages-Waiting: no Message-Account: sip:asterisk@192.168.1.2 Voice-Message: 0/0 (0/0) -- I am guessing this is some apparently unrelated and undocumented side effect of a setting I have changed recently but have not been successful chasing it down so far. I have this happening on two different servers now. -- Brian Wilson Wildsong 707-827-0001 -- _ -- 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 admin interface
You are correct. Is there any open source application in that? > According to WikiPedia, there are open-source implementations of vi > available: > All my instincts say "No no! Use emacs not vi!" but I think the OP might not know saying "vi" is intended as a joke? For small systems using a text editor is okay... watch out for typos that will disable your entire pbx... do "dialplan reload" when you change extensions.conf and then scroll back looking for ERROR messages. I find the color coding to be extremely helpful. Likewise open "asterisk -r" immediately after starting asterisk and watch for error messages. Be warned that sometimes the errors will lead you far far astray. Usually they are useful. Brian -- _ -- 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 11 on Centos: Voicemail crashes when recording message
version of 13.7 at the weekend, can't do it before > then as it's our production office system... Everything apart from VM seems > to work (although if anyone can shed any light on the frequent > "res_rtp_asterisk.c:4264 ast_rtp_read: RTP Read too short" warnings I'm > seeing, that'd also be appreciated. > > Oh - one more thing, I had to disable 2 codecs (lpc10 and ilbc) because > they used an instruction that doesn't exist on the server (it's an oldish > HP mini-server). I'm guessing from the above message that VM might be > afflicted by the same issue. Presumably compiling from source will solve > this? (I've compiled 13.7, no errors reported, but I've not tried running > it yet) > > Cheers! > Ade. > > -- > _ > -- 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 > -- Brian Wilson, GISP Wildsong 707-827-0001 -- _ -- 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] cannot find -lasteriskssl
I ran into that (and reported it) and found the easiest hack is to copy the library manually into place, when you hit that error it's built and waiting in main/ For example sudo cp main/libasteriskssl.so.1 /usr/lib cd /usr/lib sudo ln -s libasteriskssl.so.1 libasteriskssl.so sudo ldconfig Then the build can complete. I can't remember if I had to do the symlink, I think it will build anyway and then it does that step in 'make install' I always try to build new asterisks on brand new vagrant virtual machines that have the minimum required packages installed, and no cruft left over from previous builds or installations. Things like this are more likely to show up that way. Brian On Thu, May 5, 2016 at 10:21 AM, Michael Ströder wrote: > Joshua Colp wrote: > > Michael Ströder wrote: > >> Joshua Colp wrote: > >>> Michael Ströder wrote: > >>>> HI! > >>>> > >>>> I'm trying to compile asterisk 13.8.2+ on openSUSE Linux but it > fails. It seems > >>>> file ./main/libasteriskssl.so.1 is present when it fails. Building > 13.7.2 works > >>>> without any problem. It fails since 13.8.0. > >>>> > >>>> $ ./bootstrap.sh > >>>> $ ./configure > >>>> $ make menuselect.makeopts;menuselect/menuselect --enable chan_ooh323 > >>>> $ make > >>>> .. > >>>> failure (see message below) > >>>> > >>>> Any hint is appreciated. Thanks in advance. > >>> Does this problem still occur in 13.9.0-rc2? > >> > >> No. Build of 13.9.0-rc2 seems to work. > >> > >> When will 13.9.0 be released? > > > > Probably a week or so I'd guess, maybe sooner. > > Ok, then I'll wait for this release. Thanks. > > Ciao, Michael. > > > -- > _____ > -- 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 > -- Brian Wilson, GISP Wildsong 707-827-0001 -- _ -- 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] PJProject Bundled Update
Clever. Some packages are missing though. Should I convey this to someone? On Thu, Mar 31, 2016 at 12:28 PM, George Joseph wrote: > > Run ./contrib/scripts/install_prereq. I think your'e missing the > python-dev package. I'll update the Wiki. > > -- _ -- 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] PJProject Bundled Update
The way I got this build to succeed last night was by using a separate pjproject, error I get with bundle is the same after applying your patches. First patch succeeds. Second patch fails in 'configure'. What I did -- I downloaded your diffs, unpacked a fresh copy of the asterisk tarball, then applied patches. Then did configure, make menuselect, make Looks like you were working with an older version of configure? I think those changes that failed are already in the release file. Build still fails with Makefile:117: recipe for target 'source/pjsip-apps/src/python/build/_pjsua.so' failed make[2]: *** [source/pjsip-apps/src/python/build/_pjsua.so] Error 1 Makefile:20: recipe for target 'pjproject' failed make[1]: *** [pjproject] Error 2 Makefile:398: recipe for target 'third-party' failed make: *** [third-party] Error 2 I did ./configure --prefix=/usr --with-imap=system --with-pjproject-bundled I am building on a Debian 8 virtual machine. configure.rej Description: Binary data -- _ -- 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] PJProject Bundled Update
I think that I worked around the first issue (libasteriskssl) last night. Should have gone to bed earlier instead. :-) Will test your patches this morning. I build on a barebones Debian 8.x virtual machine - not "older" unless you consider "stable" = "older". On Thu, Mar 31, 2016 at 8:57 AM, George Joseph wrote: > > As you know, the ability to use a bundled version of pjproject was > introduced with Asterisk 13.8.0. > > More info on the Asterisk Wiki > <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled> > and > in this email thread > <http://lists.digium.com/pipermail/asterisk-users/2016-March/288685.html>. > > Since then I've fixed a few issues related to older versions of Debian and > CentOS which you can in these 2 patches. > https://gerrit.asterisk.org//2516 > https://gerrit.asterisk.org/2449 > > Any other feedback? I'd like to get an idea of how many folks have tried > it. > > -- Brian Wilson, GISP Wildsong 707-827-0001 -- _ -- 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
