Re: [asterisk-users] Voicemail: Tell external number instead of internal number
On Sat, Jun 16, 2012 at 4:23 PM, Stefan at WPF stefan.at@googlemail.com wrote: Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like number 1000 not available, however it should say number 123456789 not available. How can I configure this? I already googled and I guess this is really easy, but I just couldn't figure out how to do this ): So thanks for any hint :-) In your voicemail.conf, configure the mailbox as 123456789 = 1234,username,emailaddy,pager,options, and not as 1005 = 1234,username,emailaddy,pager,options And then in your extensions.conf you would call the Voicemail app like so: exten = 1005,1,Verbose(Incoming call to 123456789 transfered to SIP phone 1005) exten = 1005,n,Dial(SIP/1005,30) exten = 1005,n,Verbose(No answer, going to voicemail for 123456789) exten = 1005,n,Voicemail(123456789@default,u) exten = 1005,n,Hangup() -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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] Help choosing the right card
Eric, I sure did. It is active for the last 18 months ever since I started having this problem which coincided with my switching to Asterisk 1.8 from 1.6.2.x where I never ever had any of the DAHDI and / or Asterisk-DAHDI problems I described before. -Vladimir On 6/17/2012 12:48 AM, Eric Wieling wrote: You have verified this by using the Asterisk's DTMF debug option? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 9:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Eric, Thank you for the suggestion. In fact the problem is with FSX channel which fails to catch some DTMF tones from a phone which places an outgoing call. Shaun's theory was a delay related to swapping. -Vladimir On 6/16/2012 7:40 PM, Eric Wieling wrote: I was assuming incoming DTMF detection. Try toneduration=250 in chan_dahdi to increase the duration of transmitted DTMF on your DAHDI channels. If that fixes it, try lowering it. I find 80 usually works with even the worst IVRs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson Sent: Saturday, June 16, 2012 7:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help choosing the right card Shaun, I respect your opinion, and the swap theory is one of the valid theories. But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. -- _ -- 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 -- _ -- 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] (no subject)
http://adamdavidson-design.com/wp-content/themes/FastTrack/rogsfv.html?ncs=mmyq.jjsjss=sys.jyscjn=gyhp-- _ -- 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] Voicemail: Tell external number instead of internal number
Ah, so, really easy ;-) Thank you very much for this :-) 2012/6/17 Warren Selby wcse...@selbytech.com On Sat, Jun 16, 2012 at 4:23 PM, Stefan at WPF stefan.at@googlemail.com wrote: Hello, I have an internal extension, e.g. 1005 which is being called from an external/public number like 123456789. Now when it comes to the spoken voicemail information it says something like number 1000 not available, however it should say number 123456789 not available. How can I configure this? I already googled and I guess this is really easy, but I just couldn't figure out how to do this ): So thanks for any hint :-) In your voicemail.conf, configure the mailbox as 123456789 = 1234,username,emailaddy,pager,options, and not as 1005 = 1234,username,emailaddy,pager,options And then in your extensions.conf you would call the Voicemail app like so: exten = 1005,1,Verbose(Incoming call to 123456789 transfered to SIP phone 1005) exten = 1005,n,Dial(SIP/1005,30) exten = 1005,n,Verbose(No answer, going to voicemail for 123456789) exten = 1005,n,Voicemail(123456789@default,u) exten = 1005,n,Hangup() -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.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
[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
[asterisk-users] Missing voicemail prompt beginning
Hello, I am using the voicemail module of asterisk. When I did some test calls from my mobile phone, sometimes the beginning of the prompt was missing, e.g. instead of something like number 12345 not available I was only hearing 345 not available. Verbose level 5 on the asterisk console didn't give me any hint on this, it only shows that playback of the prompt started correctly in every test case. Any hints on how I can debug this? I think it's some problem on my local configuration, I doubt it's a problem with my SIP provider or mobile phone provider, they are both very reliable (Sipgate and T-Mobile). Thanks for any hint! Best regards Stefan -- _ -- 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] Missing voicemail prompt beginning
Stefan at WPF wrote:
beginning of the prompt was missing
User answer(500) or wait(1) before the audio prompts.
Example:
exten = s,1,Answer(500)
exten = s,n,Voicemail({$ARG1}@sip,u)
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
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deserve neither Liberty nor Safety.
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Re: [asterisk-users] Missing voicemail prompt beginning
First of all, thank you for your reply, however I see two problems with
this solution:
1) I think sometimes even more than a second from the beginning of the
prompt is missing, so I have to set a larger value, meaning in cases where
nothing of the prompt was missing, the calling person listens to a pause of
some seconds.
2) Your solutions handles the symptoms of the problem, I'd like to fix the
root cause of this problem.
Any ideas on number 2, fixing / finding the root cause of this problem?
Thanks :-)
2012/6/17 Doug Lytle supp...@drdos.info
Stefan at WPF wrote:
beginning of the prompt was missing
User answer(500) or wait(1) before the audio prompts.
Example:
exten = s,1,Answer(500)
exten = s,n,Voicemail({$ARG1}@sip,u)
exten = s,n,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
--
__**__**_
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Re: [asterisk-users] Missing voicemail prompt beginning
Stefan at WPF wrote: 2) Your solutions handles the symptoms of the problem, I'd like to fix the root cause of this problem. The root cause of the problem (Most likely) is that the channel hadn't be answered. A wait, allows the channel to be established and audio to pass. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Missing voicemail prompt beginning
2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: 2) Your solutions handles the symptoms of the problem, I'd like to fix the root cause of this problem. The root cause of the problem (Most likely) is that the channel hadn't be answered. A wait, allows the channel to be established and audio to pass. Which end do you mean with channel not answered? The asterisk end or mobile phone end of the channel? Also I am confused that it sometimes work and sometimes it does not? ): I tried with Wait(1) and Answer(1000), unfortunately both didn't change things - sometimes the complete prompt is there, sometimes the beginning is missing ): Are there any relevant logs for these things / how to check what the problem is without trying settings? Thanks :-) Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- 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-**usershttp://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] Help choosing the right card
Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. /Benny -- _ -- 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] Missing voicemail prompt beginning
Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Missing voicemail prompt beginning
Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- 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-**usershttp://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
[asterisk-users] New to Asterisk
Greetings, I am interested in learning more ablout Asterisk. Is there a recommended link for getting started. Can I set up an Asterisk server on my Win 7 local host ?? Is this what I need to do or is there another way of becoming familiar with the Asterisk product ? Any help and guidance for a new user is much appreciated ? Jim -- _ -- 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] HP DL360 G5 better than HP DL360 G7 ?
On Wed, Jun 13, 2012 at 9:06 PM, Andrew Joakimsen joakim...@gmail.com wrote: Make sure you have installed Proliant Support Pack (PSP) then you can monitor the system through HP System Management Homepage (SMH) HP publishes drivers for the network cards. I've never used them as the built in drivers seem to work, but worth a shot. Maybe included in the PSP? Also check the newest HP firmware DVD, as well as any supplemental firmware updates e.g. (check your system for compatibility first!) HP Broadcom Online Firmware Upgrade Utility for Linux x86_64 ver 2.5.14 4 Jun 2012. Thanks! We are checking and applying firmware updates. And we will look at the driver versions. I also noticed that the G7 boxes are running CentOS 5.8 vs CentOS 5.7 on the G5s. I'll rule that out by upgrading one of the G5s to 5.8 We've not had issues between CentOS releases in the past. I'm still running Asterisk 1.8.7.0 as I noticed that 1.8.12.2 would occasionally crash (and get restarted by safe_asterisk) after 10s of thousands of calls. What's the best way to make sure at least a core file gets created? Any other tips on getting crash info? It is not reproducible on demand other than putting the box into production and waiting for 50-100K calls. Hardware enumeration printout is on its way too for the original request. -- _ -- 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] New to Asterisk
Hello http://www.voip-info.org/wiki/view/Asterisk I prefer asterisk under linux sistem works better. Regards On Sun, Jun 17, 2012 at 12:28 PM, Jim Schultz jimschultz...@gmail.comwrote: Greetings, I am interested in learning more ablout Asterisk. Is there a recommended link for getting started. Can I set up an Asterisk server on my Win 7 local host ?? Is this what I need to do or is there another way of becoming familiar with the Asterisk product ? Any help and guidance for a new user is much appreciated ? Jim -- _ -- 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] Help choosing the right card
On Sun, Jun 17, 2012 at 04:39:55PM +0200, Benny Amorsen wrote: Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. This is what the patch on DAHLIN-241 [1] is intended to do (only if Asterisk is run in the real-time priority class) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Missing voicemail prompt beginning
Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com wrote: Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Help choosing the right card
On Sun, 17 Jun 2012, Shaun Ruffell wrote: What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. Any chance there is a USB or 'green' disk drive going to sleep anywhere? -- 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] Missing voicemail prompt beginning
Thank you Warren, I will temporarily skip this step, as I don't have the problem anymore, though I don't know why (for that and learning purposes the logs maybe would be still useful). I found some different settings for Asterisk and Sipgate (actually I found the settings for private users on the Sipgate website, before that I found the settings for business customers and assumed there wouldn't be a difference). When I had the problem, my sip.conf looked like this: [general] port=5060 bindaddr=0.0.0.0 context=other language=de register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=peer context=from_external_voip_provider username=SIPID defaultuser=SIPID fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes Now my sip.conf looks like this (source: http://www.sipgate.de/faq/index.php?do=displayArticlearticle=540id=257): (I have commented the additions / changes) [general] port=5060 bindaddr=0.0.0.0 context=other language=de qualify=no ; added disallow=all ; added allow=alaw ; added allow=ulaw ; added allow=g729 ; added allow=gsm; added allow=slinear; added srvlookup=yes; added register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=friend ; changed from peer to friend context = from_external_voip_provider username=SIPID ;defaultuser=SIPID ; removed fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes canreinvite=no ;added dtmfmode=rfc2833 ;added The dialplan in both cases was this: [from_external_voip_provider] exten = SIPID,1,Answer(1000) exten = SIPID,n,VoiceMail(some_number,u) exten = SIPID,n,Hangup() (I left out the Dial command for testing purposes after I found the voicemail prompt problems) If anyone has an idea why it now works without problems, please let me know for learning purposes. I still have to read up on the options. When I have more time I will probably also set the old settings again to learn how I could have identified the problem. 2012/6/17 Warren Selby wcse...@selbytech.com Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com wrote: Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- 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-**usershttp://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 -- _ -- 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:
Re: [asterisk-users] Missing voicemail prompt beginning
Sorry for the second mail, about the infrastructure: phone - asterisk - HW firewall including NAT - Sipgate SIP Provider About Software: Asterisk 1.8.13.0 running on Raspbian Debian Linux (http://www.raspbian.org/, Raspbian includes up to date Asterisk paackages while the normal Raspberry Pi Debian does not) running on a Raspberry Pi http://www.raspberrypi.org/:-) 2012/6/17 Stefan at WPF stefan.at@googlemail.com Thank you Warren, I will temporarily skip this step, as I don't have the problem anymore, though I don't know why (for that and learning purposes the logs maybe would be still useful). I found some different settings for Asterisk and Sipgate (actually I found the settings for private users on the Sipgate website, before that I found the settings for business customers and assumed there wouldn't be a difference). When I had the problem, my sip.conf looked like this: [general] port=5060 bindaddr=0.0.0.0 context=other language=de register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=peer context=from_external_voip_provider username=SIPID defaultuser=SIPID fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes Now my sip.conf looks like this (source: http://www.sipgate.de/faq/index.php?do=displayArticlearticle=540id=257): (I have commented the additions / changes) [general] port=5060 bindaddr=0.0.0.0 context=other language=de qualify=no ; added disallow=all ; added allow=alaw ; added allow=ulaw ; added allow=g729 ; added allow=gsm; added allow=slinear; added srvlookup=yes; added register = SIPID:SIP_PASS@sipgate.de/SIPID [sipgate] type=friend ; changed from peer to friend context = from_external_voip_provider username=SIPID ;defaultuser=SIPID ; removed fromuser=SIPID secret=SIP_PASS host=sipgate.de fromdomain=sipgate.de qualify=yes insecure=invite nat=yes canreinvite=no ;added dtmfmode=rfc2833 ;added The dialplan in both cases was this: [from_external_voip_provider] exten = SIPID,1,Answer(1000) exten = SIPID,n,VoiceMail(some_number,u) exten = SIPID,n,Hangup() (I left out the Dial command for testing purposes after I found the voicemail prompt problems) If anyone has an idea why it now works without problems, please let me know for learning purposes. I still have to read up on the options. When I have more time I will probably also set the old settings again to learn how I could have identified the problem. 2012/6/17 Warren Selby wcse...@selbytech.com Please excuse the top post, I'm on my phone. Before we have a better idea of what's going on, please provide the dialplan snippet that the call is using as well as the cli logs of the calls where you hear the whole prompt and where you only hear part of the prompt. Also, if you can clarify the infrastructure setup as well, that would be helpful. Thanks, --Warren Selby, dCAP On Jun 17, 2012, at 11:25 AM, Stefan at WPF stefan.at@googlemail.com wrote: Hmm, I tried calling myself (the asterisk voicemail) from another SIP provider, same problem. What always works reliable is using and calling the voicemail of my SIP Provider (Sipgate) from my mobile phone, I reliably hear the complete prompt. Doesn't this contradict the assumption that the problem is on the mobile phone side? 2012/6/17 Doug Lytle supp...@drdos.info Stefan at WPF wrote: Which end do you mean with channel not answered? The asterisk The Asterisk side. If the answer didn't fix the issue, then my guess is that it's on the cellular provider's side (Which isn't unheard of). Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__** _ -- 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-**usershttp://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
Re: [asterisk-users] Help choosing the right card
Steve, The systems I tested on are all old Dell Dimension systems with plain old PATA. I disabled all power saving features in the BIOS. -Vladimir On 6/17/2012 12:57 PM, Steve Edwards wrote: On Sun, 17 Jun 2012, Shaun Ruffell wrote: What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. Any chance there is a USB or 'green' disk drive going to sleep anywhere? -- _ -- 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] Help choosing the right card
Benny, Thank you for clarification. I did not know executable pages would not swap. Shaun wrote a patch with mlockall() and it crashed my system badly. Is there a way to lock just a specific module in memory vs. the whole Asterisk application? -Vladimir On 6/17/2012 9:39 AM, Benny Amorsen wrote: Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. /Benny -- _ -- 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] Help choosing the right card
On 6/17/2012 12:06 PM, Shaun Ruffell wrote: On Sun, Jun 17, 2012 at 04:39:55PM +0200, Benny Amorsen wrote: Vladimir Mikhelson v...@mikhelson.com writes: But interestingly enough, yesterday morning I had zero (0) bytes in the swap file and still experienced missing DTMF detection on an outgoing call. Executables do not get written to swap, their pages just get discarded under pressure, and reloaded directly from their original location on disk. The only way to ensure that Asterisk always stays in memory is to use the mlockall() system call; doing that would require patching Asterisk. This is what the patch on DAHLIN-241 [1] is intended to do (only if Asterisk is run in the real-time priority class) [1] https://issues.asterisk.org/jira/browse/DAHLIN-241 What I feel is the important clue in this case is the problem, as reported, only occurs after this system has been idle for awhile. I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. Shaun, if I understand the numbers correctly i still cannot use the patch as 562,268KB 512MB -Vladimir -- _ -- 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] Help choosing the right card
On Sun, Jun 17, 2012 at 03:43:35PM -0500, Vladimir Mikhelson wrote: On 6/17/2012 12:06 PM, Shaun Ruffell wrote: I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. Shaun, if I understand the numbers correctly i still cannot use the patch as 562,268KB 512MB You are correct. You will not be able to lock all the memory and one of the allocations will fail if you're autoloading all modules. So if you want to avoid delays incurred when the system needs to page in code pages on events, you will either need to add more memory or limit the modules that are loaded. For example, when I disable autoloading, and only load a few modules needed for a basic system only 153M is needed: # asterisk -rx 'module show' Module Description Use Count pbx_config.so Text Extension Configuration 0 res_timing_dahdi.soDAHDI Timing Interface 0 chan_dahdi.so DAHDI Telephony Driver w/PRI 0 chan_sip.soSession Initiation Protocol (SIP)0 app_dial.soDialing Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_originate.so Originate call 0 app_meetme.so MeetMe conference bridge 0 codec_ulaw.so mu-Law Coder/Decoder 0 format_sln.so Raw Signed Linear Audio support (SLN)0 format_sln16.soRaw Signed Linear 16KHz Audio support (S 0 format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 codec_alaw.so A-law Coder/Decoder 0 func_callerid.so Party ID related dialplan functions (Cal 0 func_version.soGet Asterisk Version/Build Info 0 16 modules loaded # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:152972 kB -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Help choosing the right card
On 6/17/2012 5:56 PM, Shaun Ruffell wrote: On Sun, Jun 17, 2012 at 03:43:35PM -0500, Vladimir Mikhelson wrote: On 6/17/2012 12:06 PM, Shaun Ruffell wrote: I just updated the patch since the memory locks weren't carried through after the fork call. When I apply the patch on the current head of the asterisk 1.8 branch and load all the asterisk modules by default: # asterisk -p # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:567268 kB You can see that just after load there is already 567MB locked. The systems on DAHLIN-241 started with 384M and were updated to 512M. Shaun, if I understand the numbers correctly i still cannot use the patch as 562,268KB 512MB You are correct. You will not be able to lock all the memory and one of the allocations will fail if you're autoloading all modules. So if you want to avoid delays incurred when the system needs to page in code pages on events, you will either need to add more memory or limit the modules that are loaded. For example, when I disable autoloading, and only load a few modules needed for a basic system only 153M is needed: # asterisk -rx 'module show' Module Description Use Count pbx_config.so Text Extension Configuration 0 res_timing_dahdi.soDAHDI Timing Interface 0 chan_dahdi.so DAHDI Telephony Driver w/PRI 0 chan_sip.soSession Initiation Protocol (SIP)0 app_dial.soDialing Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_originate.so Originate call 0 app_meetme.so MeetMe conference bridge 0 codec_ulaw.so mu-Law Coder/Decoder 0 format_sln.so Raw Signed Linear Audio support (SLN)0 format_sln16.soRaw Signed Linear 16KHz Audio support (S 0 format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 codec_alaw.so A-law Coder/Decoder 0 func_callerid.so Party ID related dialplan functions (Cal 0 func_version.soGet Asterisk Version/Build Info 0 16 modules loaded # cat /proc/`pidof asterisk`/status | grep VmLck VmLck:152972 kB Shaun, would it be possible to lock specific modules in RAM vs. the who;e Asterisk application? -Vladimir -- _ -- 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] Help choosing the right card
On Sun, Jun 17, 2012 at 06:14:07PM -0500, Vladimir Mikhelson wrote: Shaun, would it be possible to lock specific modules in RAM vs. the who;e Asterisk application? It is possible but not without more work. Asterisk would need to parse the output of the memory map in /proc/pid/maps and figure out where the modules are mapped into the current process' address space and then lock only those pages. Also, this would require knowing exactly which modules are needed at first. Since Asterisk really should be run in a soft real-time fashion, I still believe it's preferrable to figure out which modules are needed and then making sure all those pages can stay resident in memory. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] CDRs do not record in asteriskcdrdb using Digium repository
Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. Seems like some config is missing. Which file is responsible for this type of config. Thanks On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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] CDRs do not record in asteriskcdrdb using Digium repository
Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 *1.8.7.0-2_centos5*asterisk-current After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 *1.8.13.0-1_centos5 * * * I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: *loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk.* *loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability.* *loader.c: Module 'cdr_mysql' could not be loaded.* * * I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. Any thoughts? Thanks On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. Seems like some config is missing. Which file is responsible for this type of config. Thanks On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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] CDRs do not record in asteriskcdrdb using Digium repository
Please correct me if I'm wrong... The current version of asterisk now is 10.x Cdr_mysql is not used anymore. Now they using odbc to connect to mysql database. Why dont you try to install asterisk using source TAR.GZ ? It will make you learned where you have to do some setting... :D. Rather difficult but fun... :D From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: 18 Juni 2012 9:29 To: isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 1.8.7.0-2_centos5asterisk-current After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 1.8.13.0-1_centos5 I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk. loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability. loader.c: Module 'cdr_mysql' could not be loaded. I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. Any thoughts? Thanks On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. Seems like some config is missing. Which file is responsible for this type of config. Thanks On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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] CDRs do not record in asteriskcdrdb using Digium repository
This is not related to Asterisk Now but simply Asterisk as provided by Digium repositories and documented in Asterisk Wiki. Source install is one way to do this but that is not the issue in question. I hope someone at Digium fixes and update the repositories to current version. On Sun, Jun 17, 2012 at 10:59 PM, Ikka Vertika ikka.vert...@mitrakreasindo.com wrote: Please correct me if I’m wrong... ** ** The current version of asterisk now is 10.x Cdr_mysql is not used anymore. Now they using odbc to connect to mysql database. ** ** Why dont you try to install asterisk using source TAR.GZ ? It will make you learned where you have to do some setting... :D. Rather difficult but fun... :D ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* 18 Juni 2012 9:29 *To:* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository ** ** Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 *1.8.7.0-2_centos5*asterisk-current ** ** After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 *1.8.13.0-1_centos5 * ** ** I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. ** ** Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: *loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk.* *loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability.* *loader.c: Module 'cdr_mysql' could not be loaded.* ** ** I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. ** ** Any thoughts? ** ** Thanks ** ** ** ** ** ** ** ** On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. ** ** Seems like some config is missing. Which file is responsible for this type of config. ** ** Thanks ** ** ** ** ** ** ** ** On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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 -- _ -- 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:
Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository
Seems like there are new instructions for installing from RPM repository which seems to be working fine and updating to proper current version of Asterisk: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-YUM%28CentOS5%2FRedHatEnterpriseLinux5%29 -Bruce On Sun, Jun 17, 2012 at 11:13 PM, Bruce B bruceb...@gmail.com wrote: This is not related to Asterisk Now but simply Asterisk as provided by Digium repositories and documented in Asterisk Wiki. Source install is one way to do this but that is not the issue in question. I hope someone at Digium fixes and update the repositories to current version. On Sun, Jun 17, 2012 at 10:59 PM, Ikka Vertika ikka.vert...@mitrakreasindo.com wrote: Please correct me if I’m wrong... ** ** The current version of asterisk now is 10.x Cdr_mysql is not used anymore. Now they using odbc to connect to mysql database. ** ** Why dont you try to install asterisk using source TAR.GZ ? It will make you learned where you have to do some setting... :D. Rather difficult but fun... :D ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Bruce B *Sent:* 18 Juni 2012 9:29 *To:* isr...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository ** ** Pinpointed the problem to do with Digium repository. When I do yum install asterisk18 system installs: asterisk18.i386 *1.8.7.0-2_centos5*asterisk-current ** ** After, when I do yum update asterisk18-* then the asterisk18-core updates: asterisk18.i386 *1.8.13.0-1_centos5 * ** ** I don't know if this is a bug in Digium repository or what but 1.8.7.0-2 should NOT show as asterisk-current. ** ** Problem is that upon update, not all packages update. So, when trying to do module load cdr_mysql this error prints: *loader.c: Module 'cdr_mysql.so' was not compiled with the same compile-time options as this version of Asterisk.* *loader.c: Module 'cdr_mysql.so' will not be initialized as it may cause instability.* *loader.c: Module 'cdr_mysql' could not be loaded.* ** ** I tried download .rpm files of asterisk18-addons.rpm, asterisk18-addons-core.rpm, and asterisk18-mysql.rpm but the asterisk18-mysql.rpm fails due to dependencies upon install. So, this seems a bit complicated but it's probably an easy fix if Digium updates the system to use all REAL current version at first install instead of needing to update right after fresh install. ** ** Any thoughts? ** ** Thanks ** ** ** ** ** ** ** ** On Sun, Jun 17, 2012 at 9:34 PM, Bruce B bruceb...@gmail.com wrote: Yes, asterisk18-addons and asterisk18-addons-mysql are installed. The file cdr_mysql.so exists and I added it to modules.conf with load = cdr_mysql.so. But the module doesn't show loaded when I do module show like cdr. ** ** Seems like some config is missing. Which file is responsible for this type of config. ** ** Thanks ** ** ** ** ** ** ** ** On Sat, Jun 16, 2012 at 4:35 PM, isr...@gmail.com wrote: Did you install the addons Yum install asterisk18-addons-mysql -Original Message- From: Duncan Turnbull dun...@e-simple.co.nz Sender: asterisk-users-boun...@lists.digium.com Date: Sun, 17 Jun 2012 08:30:00 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository -- _ -- 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:
Re: [asterisk-users] Help choosing the right card
On 6/17/2012 6:21 PM, Shaun Ruffell wrote: On Sun, Jun 17, 2012 at 06:14:07PM -0500, Vladimir Mikhelson wrote: Shaun, would it be possible to lock specific modules in RAM vs. the who;e Asterisk application? It is possible but not without more work. Asterisk would need to parse the output of the memory map in /proc/pid/maps and figure out where the modules are mapped into the current process' address space and then lock only those pages. Also, this would require knowing exactly which modules are needed at first. Since Asterisk really should be run in a soft real-time fashion, I still believe it's preferrable to figure out which modules are needed and then making sure all those pages can stay resident in memory. Shaun, Thank you for the reply. I would suggest to move this conversation to JIRA as we digressed from the original topic a lot. https://issues.asterisk.org/jira/browse/DAHLIN-241?focusedCommentId=193920page=com.atlassian.jira.plugin.system.issuetabpanels%3Acomment-tabpanel#comment-193920 Thank you, Vladimir -- _ -- 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
