Re: [asterisk-users] Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction if I have callprogress=no in chan_dahdi.conf. If I change to callprogress=yes then the audio returns. My chan_dahdi.conf file is listed below. Can anyone point-out why callprogress=no isn't working? I'm assuming your telco doesn't support line reversal on answer, you need to set answeronpolarityswitch=no Hope that helps -- _ -- 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] voice quality measurement using dahdi_monitor
hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice quality so that i can talk with provider that i am getting % of voice quality.i am sure there is some better way to solve or debug .raw file and taking a decision. let me help please to solve and finding problem of voice quality. regards Dhaval -- _ -- 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] PRI voice optimization
It seems to be you are using Sangoma T1/E1 card with echo cancellation. If I am not wrong there is a parameter for echo cancel in the card configuration, try disabling that because already you have enabled echo cancel in dahdi file. Hope it help.:) On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.comwrote: Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any voice related parameter that we need to set for INDIA specific region and is ther any voice hardware tester for PRI that we can use and tell us our PRI [telco] provider that problem is not from our side. let give some idea . below are my configuration as well. # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Global data loadzone= in defaultzone = in span = 1,0,0,ccs,hdb3 bchan = 1-15 dchan = 16 bchan = 17-31 span = 2,0,0,ccs,hdb3 bchan = 32-46 dchan = 47 bchan = 48-62 span = 3,0,0,ccs,hdb3 bchan = 63-77 dchan = 78 bchan = 79-93 span = 4,0,0,ccs,hdb3 bchan = 94-108 dchan = 109 bchan = 110-124 [channels] language=en context=from-pstn switchtype=euroisdn pridialplan=local prilocaldialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes resetinterval=never rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 0 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 -- _ -- 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 -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voice quality measurement using dahdi_monitor
Hi, The question is can you record the audio to evaluate its quality? There is intrusive approach when you have a reference file that you can test against the recorded audio, or non-intrusive approach, which allows you evaluate voice quality of any call recording (no reference needed). Both correspond to ITU-T standards: P.862 for intrusive and P.563 for non-intrusive, or to Sevana AQuA (for intrusive) and Sevana NIQA (for non-intrusive). The difference is that all ITU-T recommendations related to voice quality measurement are quite expensive and involve annual royalties, but they are recognized standards. Sevana products are not recognized standards, but are used by many happy customers doing call quality assessment in VoIP, PSTN and mobile networks. Welcome to our web site: http://www.sevana.fi for further information and customer references (many are Asterisk owners) or just contact us directly. Best Regards, Sevana Oy Finland - Original Message - From: DHAVAL INDRODIYA To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 04, 2011 12:53 PM Subject: [asterisk-users] voice quality measurement using dahdi_monitor hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice quality so that i can talk with provider that i am getting % of voice quality.i am sure there is some better way to solve or debug .raw file and taking a decision. let me help please to solve and finding problem of voice quality. regards Dhaval -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voice quality measurement using dahdi_monitor
Am 04.02.2011 10:53, schrieb DHAVAL INDRODIYA: hi group , i am working on dahdi_monitor for measuring voice quality , so i want to know that on which data i can tell that this PRI lines are working properly, is there any measurement on basis of that i can make MOS. i am working from last 2-3 days but i only get idea about making .raw file and making .wav file and visulal mode of RX and TX of PRI line. what i want is measurement of voice quality so that i can talk with provider that i am getting % of voice quality.i am sure there is some better way to solve or debug .raw file and taking a decision. let me help please to solve and finding problem of voice quality. Show us: /etc/init.d/wanrouter hwprobe -- _ -- 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] PRI voice optimization
Hi Gopal, i am using *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V* card with tata PRI lines. regards dhaval On Fri, Feb 4, 2011 at 3:23 PM, Gopalakrishnan A.N sai...@gmail.com wrote: It seems to be you are using Sangoma T1/E1 card with echo cancellation. If I am not wrong there is a parameter for echo cancel in the card configuration, try disabling that because already you have enabled echo cancel in dahdi file. Hope it help.:) On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any voice related parameter that we need to set for INDIA specific region and is ther any voice hardware tester for PRI that we can use and tell us our PRI [telco] provider that problem is not from our side. let give some idea . below are my configuration as well. # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Global data loadzone= in defaultzone = in span = 1,0,0,ccs,hdb3 bchan = 1-15 dchan = 16 bchan = 17-31 span = 2,0,0,ccs,hdb3 bchan = 32-46 dchan = 47 bchan = 48-62 span = 3,0,0,ccs,hdb3 bchan = 63-77 dchan = 78 bchan = 79-93 span = 4,0,0,ccs,hdb3 bchan = 94-108 dchan = 109 bchan = 110-124 [channels] language=en context=from-pstn switchtype=euroisdn pridialplan=local prilocaldialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes resetinterval=never rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 0 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 -- _ -- 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 -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com sip%3asai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI voice optimization
I discussed this with sangoma support in the past. Sangoma says, it is NOT recommended to disable echo cancellation there. Am 04.02.2011 10:53, schrieb Gopalakrishnan A.N: It seems to be you are using Sangoma T1/E1 card with echo cancellation. If I am not wrong there is a parameter for echo cancel in the card configuration, try disabling that because already you have enabled echo cancel in dahdi file. Hope it help.:) On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any voice related parameter that we need to set for INDIA specific region and is ther any voice hardware tester for PRI that we can use and tell us our PRI [telco] provider that problem is not from our side. let give some idea . below are my configuration as well. # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order # Global data loadzone = in defaultzone = in span = 1,0,0,ccs,hdb3 bchan = 1-15 dchan = 16 bchan = 17-31 span = 2,0,0,ccs,hdb3 bchan = 32-46 dchan = 47 bchan = 48-62 span = 3,0,0,ccs,hdb3 bchan = 63-77 dchan = 78 bchan = 79-93 span = 4,0,0,ccs,hdb3 bchan = 94-108 dchan = 109 bchan = 110-124 [channels] language=en context=from-pstn switchtype=euroisdn pridialplan=local prilocaldialplan=local signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes relaxdtmf=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes resetinterval=never rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group = 0 channel = 1-15 channel = 17-31 channel = 32-46 channel = 48-62 channel = 63-77 channel = 79-93 channel = 94-108 channel = 110-124 -- _ -- 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 voicemail and centos 5
i have installed asterisk 1.8 following this doc http://www.asterisk.org/downloads/yum i installed the package asterisk18-voicemail-imapstorage-1.8.2.2-1_centos5 in order to store voicemail in imap but the application voicemail is not available when i type core show application ? in the asterisk log file i have these messages [Feb 3 19:00:20] WARNING[14311] loader.c: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Feb 3 19:00:20] WARNING[14311] loader.c: Module 'app_voicemail_imapstorage.so' could not be loaded. does someone know how to solve that problem? grocanar Newsterisk Posts: 1 Joined: Thu Feb 03, 2011 3:44 pm -- _ -- 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] Email alerts for trunks (peers)
On Fri, 2011-02-04 at 15:43 +1000, Ryan Tucker wrote: Hey Guys, I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx sip show peers' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it? -- Just a thought... I presume, correct me if i'm wrong, that if a peer goes down, it will be completely unreachable, and not just the sip-part? If so, i would have a look at the general management tools, like OpenNMS Theses can monitor the presence, by means of ICMP, and can send alerts. If not, you can still use OpenNMS, but you have to write a simpel (..) script, like the line above (with some glue) so you can either read asterisk out by means of snmp, or generate snmp-traps when a peer fails. (perhaps these already exists?) -- _ -- 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] Email alerts for trunks (peers)
On Fri, Feb 04, 2011 at 03:43:00PM +1000, Ryan Tucker wrote: I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx sip show peers' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it? What is messy about it? An alternative would be to get the same info by using AMI (using a persistent connection if you'd like), but if you go for the AMI way you could implement an event listener for PeerStatus changes: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Events -- Daniel Tryba -- _ -- 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] PRI voice optimization
Posts untopped. On Fri, Feb 4, 2011 at 11:11 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any voice related parameter that we need to set for INDIA specific region and is ther any voice hardware tester for PRI that we can use and tell us our PRI [telco] provider that problem is not from our side. let give some idea . below are my configuration as well. snip config On Fri, Feb 4, 2011 at 3:23 PM, Gopalakrishnan A.N sai...@gmail.com wrote: It seems to be you are using Sangoma T1/E1 card with echo cancellation. If I am not wrong there is a parameter for echo cancel in the card configuration, try disabling that because already you have enabled echo cancel in dahdi file. Hope it help.:) Hi Gopal, i am using Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V card with tata PRI lines. regards dhaval Dhavel, The TE410P doesn't have echo cancellation built in, do you have the VPMOCT128 Echo Cancellation module attached? The TE412P is the model with/Echo Cancellation hardware. William Stillwell -- _ -- 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] MP3 Crashing Asterisk
Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Its the way some files are encoded. Is there a way I can make it skip the files that can be played? I use the Playback() and Background() Applications (Not MP3Player) Has anyone experienced this before? I searched the archives but the few posts are all for way back in 2003, so they are not so helpful. I also tried converting the files to wav or sln but there is severe music quality loss. Anyone knows a relieable way of converting the files? Thank you! Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues and Agent penalty - how to go to second best agent when the first does not answer
I am a little confused as to what the OP wants the system to do? Call the proper agent, but when they don't answer, on the next call, it shouldn't call the same agent? OK, but for how long? 5 minutes? Until they manually unpause (current option as described by Kevin), 30 minutes? Should it then up their penalty? For how long? I should have been more precise. I don't actually expect all this to happen, but here's what I wish it did: 1) Ring agents in Round Robin fashion, but always in the same order (could simply use the already existing penalty value) 2) Always start from the top (taking into account the ringinuse value) Basically, a simple _pre-ordered_ Roundrobin. I could make this even better by (as you hinted at yourself) by using autopause and asking for an autounpause after x minutes feature. But those two things above would be wonderful, and I was actually surprised that it wasn't a possible setting. Unless I can order the agents somehow, but I seem to understand that dynamic agents are sequenced in the order in which they joined the queue, not according to some easily defined position value. How I would envision this being configured? A queue setting that would define how it handles penalty. Either in the current Ring the best agent(s) over and over again or try the good agents first, but then move on. Just a yes/no value. Mike -- _ -- 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] MP3 Crashing Asterisk
On Friday 04 Feb 2011, Timothy Smith wrote: Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Some distros used to use mpg321 instead of mpg123 (early versions of which used to suffer from non-free licence restrictions, but newer versions are LGPL) and the installer created a symbolic link so it could be invoked as mpg123. This was known to cause problems for Asterisk, which preferred the original mpg123. Try running $ mpg123 with no arguments, and note the author's name which appears in the output. If you see Michael Hipp, then it really is mpg123. If you see Joe Drew then this is really mpg321. For confirmation try $ ls -l /usr/bin/mpg123 If you see a symbolic link (cyan and permissions start with lower-case l) then this is the problem. You can always build the proper mpg123 from the Source Code (if you aren't used to doing this, you may have to install the -devel versions of any packages which you have installed but the configure script thinks you haven't, is all). When you run `make install` it probably will install itself in /usr/local/bin/mpg123 . Most distros have a default path set to look in /usr/local/bin/ before looking in /usr/bin/ ; but if you really want to make sure, then you can just copy the binary over the top of the existing symbolic link; # cp /usr/local/bin/mpg123 /usr/bin/ You might need to repeat this step last if you ever re-install mpg321 from an RPM package. -- AJS Answers come *after* questions. -- _ -- 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] Outgoing agent´s calls
I found this solution... In every line that Agent want to make an outgoing call, this call is routed by my softswitch to Asterisk, in dialplan using func_odbc.conf I could know if there any agent logged in this line because I have this information in my DB. Then I set accoutncode field from CDR with the agent id. If there aren´t any agent logged in this line I reject the call. Thanks! On Thu, Feb 3, 2011 at 12:02 PM, Danny Nicholas da...@debsinc.com wrote: Then DISA (I had it as DASI in OP because I’m working from not so good memory) is probably your best bet. It is a simple built-in feature that let’s you get an access code in the dialplan before performing an action such as dialing. Check this link http://nerdvittles.com/index.php?p=73 -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Thursday, February 03, 2011 6:14 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Outgoing agent´s calls Yes, my agents dial “willy-nilly”... I can´t use the ex-girlfriend because, the line numbers that uses the agents are diferent. May be agent 1 today use line number 553455 and tomorrow 553461... On Wed, Feb 2, 2011 at 3:51 PM, Danny Nicholas da...@debsinc.com wrote: -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *equis software *Sent:* Wednesday, February 02, 2011 12:26 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Outgoing agent´s calls Hi, is there any way to manage outgoing calls from agents? Mi agents are answering in pstn lines. I can send agents outgoing calls to my Asterisk but I don't know wich agent is making the call...because, may be he is unregister... Is there any solution? Thanks You could start with DASI and ex-girlfriend logic in your dialplan. I’m assuming now that your agents dial “willy-nilly” (with no restrictions and you find out what they did when you read the CDR). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] MP3 Crashing Asterisk
On Fri, 4 Feb 2011, Timothy Smith wrote: I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Unfortunately the logs do not give me a clear fault or cause of crash but i can clearly see that ts because of the MP3 files. Read up on how to create a crash dump and submit a bug report. A 'bad' file shouldn't crash Asterisk. I also tried converting the files to wav or sln but there is severe music quality loss. By converting to MP3, some would say the 'music quality' has already been lost :) I convert MP3s with the following: mpg123 -q -w example.mp3.wav example.mp3 sox example.mp3.wav -c 1 -s -w -r 8000 example.wav normalize example.wav If this doesn't help, can you post links to the MP3 and the WAV? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MP3 Crashing Asterisk
Thank you for the pointers. I have checked my system, I seem to have the real mpg123. see below. -- [root@ivr2 en]# mpg123 You made some mistake in program usage... let me briefly remind you: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.13.0; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes . . . See the manpage mpg123(1) or call mpg123 with --longhelp for more parameters and information. [root@ivr2 en]# ls -l /usr/bin/mpg123 ls: cannot access /usr/bin/mpg123: No such file or directory [root@ivr2 en]# which mpg123 /usr/local/bin/mpg123 [root@ivr2 en]# ls -l /usr/local/bin/mpg123 -rwxr-xr-x. 1 root root 386286 Dec 15 00:13 /usr/local/bin/mpg123 [root@ivr2 en]# I also think I installed it using yum, however, i can still install a version from sources, just to be sure. Could you please give me the exact URLwhere I can download a version that works well with asterisk? Thank alot! Tim On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: On Friday 04 Feb 2011, Timothy Smith wrote: Hi Users, I have a problem with some of my mp3 files. they crash the system (Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to play them. Some distros used to use mpg321 instead of mpg123 (early versions of which used to suffer from non-free licence restrictions, but newer versions are LGPL) and the installer created a symbolic link so it could be invoked as mpg123. This was known to cause problems for Asterisk, which preferred the original mpg123. Try running $ mpg123 with no arguments, and note the author's name which appears in the output. If you see Michael Hipp, then it really is mpg123. If you see Joe Drew then this is really mpg321. For confirmation try $ ls -l /usr/bin/mpg123 If you see a symbolic link (cyan and permissions start with lower-case l) then this is the problem. You can always build the proper mpg123 from the Source Code (if you aren't used to doing this, you may have to install the -devel versions of any packages which you have installed but the configure script thinks you haven't, is all). When you run `make install` it probably will install itself in /usr/local/bin/mpg123 . Most distros have a default path set to look in /usr/local/bin/ before looking in /usr/bin/ ; but if you really want to make sure, then you can just copy the binary over the top of the existing symbolic link; # cp /usr/local/bin/mpg123 /usr/bin/ You might need to repeat this step last if you ever re-install mpg321 from an RPM package. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [newbie] Conference call
On Fri, 4 Feb 2011 10:54:56 +0330, Pezhman Lali l...@lopl.net wrote: Meetme is a default conference application, but you can try conference or konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Conference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konference http://www.voip-info.org/wiki/view/Asterisk+cmd+Konferencethe installation for conference or konference are more easy Thanks for the links. I'll read up on Conference/Konference. BTW, am I correct in understanding that using Flash() in the dialplan is the programmatic equivalent of the flash hook (R key on European handsets) to put someone on hold and dialing a second call? What about combining the two calls into a conference call? Thank you. -- _ -- 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] MP3 Crashing Asterisk
(Putting everything back into the right order, and stripping out unnecessary bits, for the sake of anybody searching the archives in future.) On Friday 04 Feb 2011, Timothy Smith wrote: On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Try running $ mpg123 with no arguments, and note the author's name which appears in the output. Thank you for the pointers. I have checked my system, I seem to have the real mpg123. see below. [root@ivr2 en]# mpg123 You made some mistake in program usage... let me briefly remind you: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.13.0; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes Hmm . That's the real mpg123 alright. [root@ivr2 en]# which mpg123 /usr/local/bin/mpg123 I also think I installed it using yum, however, i can still install a version from sources, just to be sure. Could you please give me the exact URLwhere I can download a version that works well with asterisk? If it's in /usr/local/bin/ then it almost certainly was built from Source Code. Our working installation (on Debian Lenny) is Asterisk 1.6.2.9 (built from source) with mpg123 version 1.4.3 (installed from a .deb). More tests to try: Can you listen to an mp3 file through the Asterisk server's own sound card (if it has one; if not, use the -w option to write to a .wav file, and test that by copying it to another machine which has a sound card), by invoking mpg123 from the command line? Try $ file $(which asterisk) $ file /usr/local/bin/mpg123 and make sure both are compiled for the same architecture (ELF 64-bit LSB executable or ELF 32-bit LSB executable). If one is 32-bit and the other is 64-bit, you *will* get problems. -- AJS Answers come *after* questions. -- _ -- 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] PRI voice optimization
On Fri, Feb 4, 2011 at 12:41 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any voice related parameter that we need to set for INDIA specific region and is ther any voice hardware tester for PRI that we can use and tell us our PRI [telco] provider that problem is not from our side. let give some idea . below are my configuration as well. If 70% of calls get good quality then chances are its not your problem for the 30%. Things to look at (for the 30%): 1. Any specific internal phones that this problem sticks with? 2. Other end a cell phone? or maybe VoIP? 3. Any bluetooth involved? Bluetooth IMHO is a disaster of a technology when it comes to realtime voice as in phone conversations. Worse than G.729. It should never be used for a professional business conversation, wired headsets for cell phones still beat any wireless solutions. For desk phones proprietary RF is far better than BT. 4. What type of call quality? A. Garbled as in under water (jitter???) or B. Echo as in hearing yourself back after some ms? or C. Static Select case case A your end or far end? case B your end or far end? case C probably far end. Hope this helps -- _ -- 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] SoftHangup on asterisk 1.8.2.3
I am trying to use SoftHangup in my dialplan, but it's either not
working or I'm not using it correctly.
when i'm on the console, i see:
pbx1*CLI core show channels
Channel Location State Application(Data)
SIP/vgw1-00a2 2156181505@inbound:1 Up AppDial((Outgoing Line))
SIP/143-009f s@macro-SaferSIPDial Up Dial(SIP/99302156181505@vgw1,,
2 active channels
1 active call
194 calls processed
pbx1*CLI
in my dialplan, i have:
exten = s,1,Set(CHAN=${SHELL(asterisk -rx core show channels | awk
'/^SIP\/vgw1-/ { print $1 }' | head -1)})
exten = s,n,SoftHangup(${CHAN})
exten = s,n,Wait(2)
When I dial the extension to invoke the above dialplan code, the console
shows:
-- Executing [s@nineoneone:10] SoftHangup(SIP/111-00a3,
SIP/vgw1-00a2) in new stack
but the SIP/vgw1-00a2 is still active. If I use 'channel request
hangup SIP/vgw1-00a2', the call is dropped instantly.
Am I using SoftHangup incorrectly?
--
Jeremy Kister
http://jeremy.kister.net./
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Re: [asterisk-users] MP3 Crashing Asterisk
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: (Putting everything back into the right order, and stripping out unnecessary bits, for the sake of anybody searching the archives in future.) Thanks! On Friday 04 Feb 2011, Timothy Smith wrote: On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Try running $ mpg123 with no arguments, and note the author's name which appears in the output. Thank you for the pointers. I have checked my system, I seem to have the real mpg123. see below. [root@ivr2 en]# mpg123 You made some mistake in program usage... let me briefly remind you: High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.13.0; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes Hmm . That's the real mpg123 alright. [root@ivr2 en]# which mpg123 /usr/local/bin/mpg123 I also think I installed it using yum, however, i can still install a version from sources, just to be sure. Could you please give me the exact URLwhere I can download a version that works well with asterisk? If it's in /usr/local/bin/ then it almost certainly was built from Source Code. Our working installation (on Debian Lenny) is Asterisk 1.6.2.9 (built from source) with mpg123 version 1.4.3 (installed from a .deb). More tests to try: Can you listen to an mp3 file through the Asterisk server's own sound card (if it has one; if not, use the -w option to write to a .wav file, and test that by copying it to another machine which has a sound card), by invoking mpg123 from the command line? Unfortunately, I cannot as the server is in a remote location. I also have to read about crash dumps to establish which file exactly cuases the crash. I have too much debugging but I usually see [Feb 5 08:15:51] WARNING[4895] mp3/interface.c: Junk at the beginning of frame 49443303 or [Feb 5 02:14:05] WARNING[7447]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304 just before the crash. Try $ file $(which asterisk) $ file /usr/local/bin/mpg123 and make sure both are compiled for the same architecture (ELF 64-bit LSB executable or ELF 32-bit LSB executable). If one is 32-bit and the other is 64-bit, you *will* get problems. I seem to have the same versions. [root@ivr ~]# file $(which mpg123) /usr/local/bin/mpg123: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.32, not stripped [root@ivr ~]# file $(which asterisk) /usr/sbin/asterisk: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.32, not stripped [root@ivr1 ~]# -- _ -- 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
