Re: [asterisk-users] Being attacked by an Amazon EC2 ...
Amazon is pretty clever! Ever seen V on TV? Amazon talks a pretty good game out of one side of their PR mouthpiece, but as a few of you note above, they abuse words like quickly and temper everything with when Amazon determines. This is a PR damage control statement. It means they are hearing the shots fired by irate server operators/owners and I say you should keep that pressure on until you actually see them acting QUICKLY and not dicking you around, asking you to resubmit reports, etc. I know some of you whose servers have been attacked. I know that you are extremely capable network admins, programmers, VoIP engineers, etc, which means your reports are technically at the same level or higher than the people at Amazon that receive them. Conclusion: Amazon is still dancing, start shooting higher then their feet. /r -- _ -- 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] Being attacked by an Amazon EC2 ...
On Tue, 20 Apr 2010, Frank Bulk wrote: Please take note of their posting: https://aws.amazon.com/security/ which discusses the issue and what they're doing to improve response. And is anyone on the list worthy of being considered a significant SIP provider to be honoured with the privilege of working with them? Gordon -- _ -- 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] Calls drop after 20 seconds
I've had this same problem at times and never found a satisfactory explanation or resolution for it but here's a couple of things you can try. Upgrade phones to latest/most stable firmware Upgrade routers to latest/most stable firmware Both these actions have worked for us at one time or another. Ish Alejandro Recarey wrote: Doug, thanks for the help, already looked it up, but it does not seem to be a NAT issue (which is what most posters suggest when googling) Danny, those are billsec durations, the call has been established and media is being passed for 20 seconds. Thanks again! Alex -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?
1. Subject.
2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in
SOURCES
3. for --without dahdi
diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec
750a750
%{_libdir}/asterisk/modules/res_timing_dahdi.so
879d878
%{_libdir}/asterisk/modules/res_timing_dahdi.so
--
_
-- 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] Calls drop after 20 seconds
--- On Mon, 4/19/10, Alejandro Recarey alexreca...@gmail.com wrote: their calls drop after 20 seconds or so. All of my customers use Grandstream GXW4004 telephony adapters. Check out the early dial feature in the Grandstream products (if you enabled it) and play with the pedantic option. You might want to take a look at this: https://issues.asterisk.org/view.php?id=14652 -- _ -- 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] Being attacked by an Amazon EC2 ...
On Apr 21, 2010, at 4:50 AM, Gordon Henderson wrote: On Tue, 20 Apr 2010, Frank Bulk wrote: Please take note of their posting: https://aws.amazon.com/security/ which discusses the issue and what they're doing to improve response. And is anyone on the list worthy of being considered a significant SIP provider to be honoured with the privilege of working with them? Gordon None of the carriers I deal with have been contacted. Of course, them only contacting significant providers... does that mean it's ok if the attacks happen to non-significant providers or end-points? ---fred http://qxork.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] Calls drop after 20 seconds
Alejandro Recarey schrieb: Doug, thanks for the help, already looked it up, but it does not seem to be a NAT issue (which is what most posters suggest when googling) Danny, those are billsec durations, the call has been established and media is being passed for 20 seconds. Thanks again! Alex Hi, How do you dial the users? direct with the peername or something like ex...@ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. best regards steve -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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] Calls drop after 20 seconds
Like the poster below said, do a sip debug on a call and see which end sends the bye message or ends the call and go from there. That should give you some sort of clue as to who is having a timer issue. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stefan Schmidt Sent: Wednesday, April 21, 2010 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls drop after 20 seconds Alejandro Recarey schrieb: Doug, thanks for the help, already looked it up, but it does not seem to be a NAT issue (which is what most posters suggest when googling) Danny, those are billsec durations, the call has been established and media is being passed for 20 seconds. Thanks again! Alex Hi, How do you dial the users? direct with the peername or something like ex...@ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. best regards steve -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // v...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - -- _ -- 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] Being attacked by an Amazon EC2 ...
On Wed, Apr 21, 2010 at 2:55 PM, Fred Posner f...@teamforrest.com wrote: On Apr 21, 2010, at 4:50 AM, Gordon Henderson wrote: On Tue, 20 Apr 2010, Frank Bulk wrote: Please take note of their posting: https://aws.amazon.com/security/ which discusses the issue and what they're doing to improve response. And is anyone on the list worthy of being considered a significant SIP provider to be honoured with the privilege of working with them? Gordon None of the carriers I deal with have been contacted. Of course, them only contacting significant providers... does that mean it's ok if the attacks happen to non-significant providers or end-points? ---fred http://qxork.com If it got to their BS/PR page/blog it means they're hearing about complaints on the net as well as people like you submitting. Everyone please keep posting where you can and sooner or later, someone big will pick up the story. Funny, I'd think the most worthy people to comment on this issue are on this list. That's the feedback they should be looking for and working on at Amazon EC2. /r -- _ -- 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] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?
I didn't know there was an RPM for centos with asterisk in it.
I personally think that's a bad idea. There are a lot of source options.
app_fax.so in particular depends on SpanDSP, and particular versions thereof.
That's probably why it's missing from somebody's RPM.
Build from source.
On Wed, Apr 21, 2010 at 7:02 AM, Самусенко Андрей samuse...@msm.ru wrote:
1. Subject.
2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in
SOURCES
3. for --without dahdi
diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec
750a750
%{_libdir}/asterisk/modules/res_timing_dahdi.so
879d878
%{_libdir}/asterisk/modules/res_timing_dahdi.so
--
_
-- 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] Being attacked by an Amazon EC2 ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Randy R wrote: On Wed, Apr 21, 2010 at 2:55 PM, Fred Posner f...@teamforrest.com wrote: On Apr 21, 2010, at 4:50 AM, Gordon Henderson wrote: On Tue, 20 Apr 2010, Frank Bulk wrote: Please take note of their posting: https://aws.amazon.com/security/ which discusses the issue and what they're doing to improve response. And is anyone on the list worthy of being considered a significant SIP provider to be honoured with the privilege of working with them? Gordon None of the carriers I deal with have been contacted. Of course, them only contacting significant providers... does that mean it's ok if the attacks happen to non-significant providers or end-points? ---fred http://qxork.com If it got to their BS/PR page/blog it means they're hearing about complaints on the net as well as people like you submitting. Everyone please keep posting where you can and sooner or later, someone big will pick up the story. Funny, I'd think the most worthy people to comment on this issue are on this list. That's the feedback they should be looking for and working on at Amazon EC2. /r We might me reading their PR wrong... Maybe there were large SIP providers that were compromised due to this attack... Maybe they are keeping that quiet at the request of those providers... It could also be that the aliens in hiding in Colorado are behind the whole thing! ... Oh no! I've said too much!!! LOL... It could actually be the case that this whole issue went beyond what we are seeing, and they are trying to protect one of their Whale customers... Needless to say, what about the SSH brute force attacks that originate from their network? What about the SPAM that flows like a fountain from their net blocks? This was nothing more then PR hype... Stu - -- For six long years I've been in trouble, no pleasure here on earth I found. For in this world I'm bound to ramble, I have no friends to help me now. -- The Soggy Bottom Boys - I am a man of constant sorrow -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iQIcBAEBCAAGBQJLzxhvAAoJEFKVLITDJSGS7boP/A00AIG02wKVejBPM+EZnqwE zc12a0RwjbS9j3LjxSbutfDUBb5LphJpknVHy1HF7pPj5Dm3LNooVhSUq8UU+vO+ iSGIGMDVij943dGKo2bInhhZmc9rCAyBmmrRn/AP/YvQ3ZxrcJPyirOQOeEpTMee m1ctlVsP2/O5M8Igv8Hm+eE4ZlDDsTSSDr3M0W80y1wMzUD/XLtEsOWexT3wVRUY WuErhbc7xZcySgEy7GsH3+O3BFhuV2JYwr0bkF+qVcdDbDL13aiBEqoJqWDOqhJI dcgY1JYra8wUU5aum/1awH+psxpx0WTsIUr34yDDUoRRCubmVjeDL4ZBVeT4O8E8 b2UvRalGhtFl8zm8FoaBCWmG5fNoorNasoyTnkyANsAnvdW72T9Wn5yWKAwVaYZe VlX7S9bcpBV880jgm6hV7rrDFizyy4Lo96f1eoSlwNy8e4LI/bp/dn5f54RBDj5k fpckpYFZFz0kAOwnAAlwKOmHgUr/jMqMMFL6ZyF/7fl7phwVKHm1DwspF0soLJkF GEAztCBRG02++eePNCpJWk/WdNzGSA6btveOSWbYy+BkZ8UTmr9IKXp2lOsBeXJa xrCv5vgB0s9TAd/QPoBRY8XLEp4BYEL9+cDzpclbMpi5ybVwviAGjjm9gNnx0Fd7 /8HHyve0W1uNIVIsHzDz =N7lH -END PGP SIGNATURE- -- _ -- 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] Being attacked by an Amazon EC2 ...
On Wed, Apr 21, 2010 at 9:23 AM, Stuart Sheldon s...@actusa.net wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Randy R wrote: On Wed, Apr 21, 2010 at 2:55 PM, Fred Posner f...@teamforrest.com wrote: On Apr 21, 2010, at 4:50 AM, Gordon Henderson wrote: On Tue, 20 Apr 2010, Frank Bulk wrote: Please take note of their posting: https://aws.amazon.com/security/ which discusses the issue and what they're doing to improve response. And is anyone on the list worthy of being considered a significant SIP provider to be honoured with the privilege of working with them? Gordon None of the carriers I deal with have been contacted. Of course, them only contacting significant providers... does that mean it's ok if the attacks happen to non-significant providers or end-points? ---fred http://qxork.com If it got to their BS/PR page/blog it means they're hearing about complaints on the net as well as people like you submitting. Everyone please keep posting where you can and sooner or later, someone big will pick up the story. Funny, I'd think the most worthy people to comment on this issue are on this list. That's the feedback they should be looking for and working on at Amazon EC2. /r We might me reading their PR wrong... Maybe there were large SIP providers that were compromised due to this attack... Maybe they are keeping that quiet at the request of those providers... It could also be that the aliens in hiding in Colorado are behind the whole thing! ... Oh no! I've said too much!!! LOL... It could actually be the case that this whole issue went beyond what we are seeing, and they are trying to protect one of their Whale customers... Needless to say, what about the SSH brute force attacks that originate from their network? What about the SPAM that flows like a fountain from their net blocks? This was nothing more then PR hype... Stu Assuming that every such spamming/hacking/attack site is funded on a stolen identity/CC number, it will soon sink into Amazon that they are getting a bad rep, and losing money on such problems, as all such charges are reversed when the identity theft is discovered... How they overcome the problem, should be a tribute to the marvelous power of human ingenuity. murf -- Steve Murphy ParseTree Corp -- _ -- 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] Being attacked by an Amazon EC2 ...
On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy m...@parsetree.com wrote: Assuming that every such spamming/hacking/attack site is funded on a stolen identity/CC number, it will soon sink into Amazon that they are getting a bad rep, and losing money on such problems, as all such charges are reversed when the identity theft is discovered... How they overcome the problem, should be a tribute to the marvelous power of human ingenuity. Interesting point about the stolen CC numbers. If that is true, then they will be forced to investigate for their own internal damage control. /r -- _ -- 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] Asterisk choking on voice messages announcements
Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Are your sound files being transcoded or played back in their native formats? On 04/21/2010 12:25 PM, bruce bruce wrote: Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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
[asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast
Hello, As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to compiling Asterisk ourselves to get the results we'd like. Any help is appreciated. Thanks Pat Davila -- http://tllts.org/ - The Linux Link Tech Show http://mythtvcast.com/ - MythTVCast http://patdavila.wordpress.com - My blog -- _ -- 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] Being attacked by an Amazon EC2 ...
Randy- On Wed, Apr 21, 2010 at 5:33 PM, Steve Murphy m...@parsetree.com wrote: Assuming that every such spamming/hacking/attack site is funded on a stolen identity/CC number, it will soon sink into Amazon that they are getting a bad rep, and losing money on such problems, as all such charges are reversed when the identity theft is discovered... How they overcome the problem, should be a tribute to the marvelous power of human ingenuity. Interesting point about the stolen CC numbers. If that is true, then they will be forced to investigate for their own internal damage control. You are nothing if not persistent, an excellent quality in a case like this. By now I'm sure Amazon execs are wondering who is this Randulo guy, hehe. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Yes, it's all g.711 ulaw. On Wed, Apr 21, 2010 at 1:37 PM, Darrick Hartman (lists) dhart...@djhsolutions.com wrote: Are your sound files being transcoded or played back in their native formats? On 04/21/2010 12:25 PM, bruce bruce wrote: Hi Everyone, I have a weired situation where calls in and out are proceessed all right but when I dial *97 Asterisk is literally choking when it comes to announcements like Password or Call from 205-456-. Each one of those announcements can take like 10+ seconds to finish with most of it not even compoundable. I run top and there is no heavy load on CPU or RAM. I dial out and it's all fine. Can you please give me some pointers as to where to look for the problem? Also, if I allow a call to go to voice-mail on my extension, the announcement, The person at extension 4000 is not available is also garbled and very slow like a choking sound. This is serious because people think they are have reached a faulty answering machine or just cut off because there is a long instance of silence sometime. Thanks -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.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] Improving audio bitrate for all callers in aconference room for a podcast
Pat- As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to compiling Asterisk ourselves to get the results we'd like. Let me understand first: the SIP phone doing the recording is not one of the people on the conference? It's in monitor mode, for recording purposes only? If that's the case, then you can't achieve audio quality higher than the individual conference node channels themselves -- sort of a 'lowest common denominator' situation. If you could get all nodes using a wideband codec (say G722), and if Asterisk supports wideband mixing and recording (i.e. everything done at 16 kHz sampling rate), then you might be able to do it. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Are you running asterisk in a virtual machine? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast
As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to compiling Asterisk ourselves to get the results we'd like. Let me understand first: the SIP phone doing the recording is not one of the people on the conference? It's in monitor mode, for recording purposes only? If that's the case, then you can't achieve audio quality higher than the individual conference node channels themselves -- sort of a 'lowest common denominator' situation. If you could get all nodes using a wideband codec (say G722), and if Asterisk supports wideband mixing and recording (i.e. everything done at 16 kHz sampling rate), then you might be able to do it. -Jeff Jeff, So the first thing to improve audio quality is to switch over to a higher quality codec like G722. What are the other higher quality codecs we can use? Everyone connecting should make sure they're using the higher quality codec? Is there any way to configure a stock Asterisk install to use wideband mixing or will we have to compile our own? Thanks again Pat -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
yes, it's on Amazon. On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock rrb3...@gmail.com wrote: Are you running asterisk in a virtual machine? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Then use a timing source if the version is correct (1.6.1 or 2), or install dahdi-dummy, which can be quite some amount of work On Wed, Apr 21, 2010 at 12:35 PM, bruce bruce bruceb...@gmail.com wrote: yes, it's on Amazon. On Wed, Apr 21, 2010 at 2:26 PM, Ryan Bullock rrb3...@gmail.com wrote: Are you running asterisk in a virtual machine? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast
As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to compiling Asterisk ourselves to get the results we'd like. Let me understand first: the SIP phone doing the recording is not one of the people on the conference? It's in monitor mode, for recording purposes only? If that's the case, then you can't achieve audio quality higher than the individual conference node channels themselves -- sort of a 'lowest common denominator' situation. If you could get all nodes using a wideband codec (say G722), and if Asterisk supports wideband mixing and recording (i.e. everything done at 16 kHz sampling rate), then you might be able to do it. -Jeff Jeff, So the first thing to improve audio quality is to switch over to a higher quality codec like G722. What are the other higher quality codecs we can use? Everyone connecting should make sure they're using the higher quality codec? Is there any way to configure a stock Asterisk install to use wideband mixing or will we have to compile our own? Thanks again Pat I found this link: http://www.voip-info.org/wiki/view/Asterisk+codecs So every client that connects to the conference would have to be configured to use whatever codec we wind up using. -- _ -- 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] Interpbx connection
Steve, You're completely right!! it seems like my colleague gave me a wrong info (probably a firewall issue), i was also curious (before i read your response) so i tried this in my network and really it has nothing to do with call setup or peer authentication, sorry for the wrong info Guys! 2010/4/19 Steve Edwards asterisk@sedwards.com Un-top-posting... 2010/4/14 khalid touati khalidtou...@gmail.com i've connecting two pbx server successfully for several times using the following config: register = USPBX:myp...@122.11.176.35uspbx%3amyp...@122.11.176.35 [PBX1] type=friend host=122.11.176.35 trunk=yes sercret=mypass context=external deny=0.0.0.0/0.0.0.0 permit=122.11.176.35/255.255.255.240 insecure=very allow=all nat=yes qualify=yes canreinvite=no in the other and it's the analog. but now i can only dial from one end, and the other en d is giving me this error. Apr 14 16:44:21 ERROR[26502]: chan_sip.c:6659 register_verify: Peer 'PBX1' is trying to register, but not configured as host=dynamic when dialing a fast busy signal and it sauys in the CLI: CONGESTION. any help please!!! -- Abdullah On Mon, 19 Apr 2010, khalid touati wrote: for people's future references: we found out that the option in DIAL application in the extensions.conf has to be the same from both side, the issue was India server was using tr while US server was using TWw so we made them both using tr and that solved the issue, i guess if one side is set to trTWw that would work regardless of the other side but didn't try though. have a headeache-free experience with asterisk the future of telephony :)! Dial() options don't have any relationship to registration failures -- they happen at different times. Registration failures may cause dial() failures. I don't understand the relationship between ringing, transfer and recording options and dial() returning congestion. I'd suggest investigating exactly which combination is causing congestion before concluding it is unrelated to the registration failure. -- 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 -- Abdullah -- _ -- 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] long return times from System() calls with 1.6.2.6?
On Thu, Apr 8, 2010 at 6:04 PM, David Backeberg dbackeb...@gmail.com wrote:
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts that do these test -e calls runs
lightning fast. It's like maybe there's something going on where it
needs to run sudo or something?
There was a big change in the way the ast_safe_system() API call (used
by the System() dialplan application) works between 1.6.0 and 1.6.2;
it's possible you are seeing a side effect of this change. If you'd like
to experiment, open up main/app.c (in 1.6.2), search for the
ast_close_fds_above_n() function, and in the for() loop that runs from
'n+1' to 'rl.rlim_cur', change 'rl.rlim_cur' to '4096'. If that changes
the behavior, we've found the culprit, and you can open an issue on
issues.asterisk.org so this can be investigated.
On further review, I'm having other problems with this machine. I need
more data points before I point the finger at asterisk, as it seems
that the other 1.6.2.6 machine was fine.
I now have another system as a datapoint. This is a second CentOS
system, and it's having the same weird lag problem when calling
System().
I'm going to try Kevin's proposal. This new system is a very beefy,
new i7 multi-cpu, multi-core, boatloads of ram box, and there's no
good explanation for lag on this idle system. Will report results
soon.
--
_
-- 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] long return times from System() calls with 1.6.2.6?
On Thu, Apr 8, 2010 at 5:01 PM, Kevin P. Fleming kpflem...@digium.com wrote:
David Backeberg wrote:
I'm doing really, really innocent things, like:
exten = s,n,System(test -e ${MESSAGE_PATH}${EXTEN})
So I did some more testing. Same dialplan, reverted to
asterisk-1.6.0.13, and the contexts that do these test -e calls runs
lightning fast. It's like maybe there's something going on where it
needs to run sudo or something?
There was a big change in the way the ast_safe_system() API call (used
by the System() dialplan application) works between 1.6.0 and 1.6.2;
it's possible you are seeing a side effect of this change. If you'd like
to experiment, open up main/app.c (in 1.6.2), search for the
ast_close_fds_above_n() function, and in the for() loop that runs from
'n+1' to 'rl.rlim_cur', change 'rl.rlim_cur' to '4096'. If that changes
the behavior, we've found the culprit, and you can open an issue on
issues.asterisk.org so this can be investigated.
Very insightful.
This totally fixed the problem.
Opening an issue on bugtracker now.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast
Pat- As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to compiling Asterisk ourselves to get the results we'd like. Let me understand first: the SIP phone doing the recording is not one of the people on the conference? It's in monitor mode, for recording purposes only? If that's the case, then you can't achieve audio quality higher than the individual conference node channels themselves -- sort of a 'lowest common denominator' situation. If you could get all nodes using a wideband codec (say G722), and if Asterisk supports wideband mixing and recording (i.e. everything done at 16 kHz sampling rate), then you might be able to do it. -Jeff So the first thing to improve audio quality is to switch over to a higher quality codec like G722. What are the other higher quality codecs we can use? Another possibility might be G711.1. Everyone connecting should make sure they're using the higher quality codec? Yes. If a few don't and a few do then you would have a couple of issues: -transcoding has to take place prior to mixing, so whatever SIP software you're using has to correctly handle negotiation and call setup and one of the software components in your setup has to do actual transcoding work on RTP (voice data) packets. Are you using Asterisk for the conferencing function? Or only recording? If the latter then who/what does the conferencing (mixing) ? -if you were to use Asterisk for transcoding, I'm not sure how Asterisk would handle that. It could downsample the wideband nodes, then you get no audio quality improvement , or it could upsample the G711 (or other nodes) and your recording would sound better when the wideband nodes are talking Is there any way to configure a stock Asterisk install to use wideband mixing or will we have to compile our own? Not sure! -Jeff -- _ -- 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] long return times from System() calls with 1.6.2.6?
issue opened. https://issues.asterisk.org/view.php?id=17223 -- _ -- 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] High Availability - Shared Database - Ideas?
I am investigating High Availability solutions for my front end servers. I got into a discussion regarding replicated local databases versus a single fiber connected shared database on an EMC. Is anyone running a shared database on a SAN? Care to comment on your findings... Thanks, Robert -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Issue With Polycom Phones
Oh, to answer the second question, I am using 3.2.3, the latest Polycom firmware. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Tuesday, April 20, 2010 4:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Odd Issue With Polycom Phones On 04/19/2010 02:22 PM, Jay Vocaire wrote: I have searched everywhere, but cannot seem to find anyone else talking about this issue. Maybe I am just using the wrong search terms. I am running Asterisk 1.6.2 and multiple Polycom phones all with 3.2.3 (the latest) firmware on them. I am having an issue with my 550's and my 6000's (but oddly enough, not my 320's). Whenever a number is dialed on hook, and then the speakerphone button is pressed, the number is dialed twice. If the handset is picked up, or the Dial softkey is pressed, the call is only sent once. This leads me to believe it is a phone issue, not a * config issue, but I have no way of telling. I can verify that there are two call started via the snippet below: SNIP The first hangup was triggered right away (without me doing anything), the second hangup was me actually hanging up the calling phone. It does the same thing if I dial an outside line. Any idea where to start trying to solve this? Has anyone else seen it, and can point me to the fix that I could not find with Google? Thanks. I would recommend that you enable debugging on the peer only and check to see if you see two invites come from the phone. Two invites with different call ID's would indicate it is indeed the phone making two calls. One would indicate that it MAY be an Asterisk issue. Are you using the latest Polycom firmware, btw? -- _ -- 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] Odd Issue With Polycom Phones
Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the tag= determines the call. The first time it sends like this: --- SIP read from UDP:x.x.x.x:5060 --- INVITE sip:3...@y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe3e15c76913F8BDD From: 3271 sip:3271@ y.y.y.y ;tag=990EE6B0-8E3DCEA7 To: sip:3261@ y.y.y.y;user=phone CSeq: 1 INVITE Call-ID: 96a1fe9c-88f06c73-7e209...@x.x.x.x Contact: sip:3271@ x.x.x.x:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: --- SIP read from UDP:x.x.x.x:5060 --- INVITE sip:3261@ y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6f7a6692AF94008 From: 3271 sip:3271@ y.y.y.y ;tag=990EE6B0-8E3DCEA7 To: sip:3261@ y.y.y.y;user=phone CSeq: 2 INVITE Call-ID: 96a1fe9c-88f06c73-7e209322@ x.x.x.x Contact: sip:3271@ x.x.x.x:5060 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3261@ y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. Thanks. -Jay -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Tuesday, April 20, 2010 4:57 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Odd Issue With Polycom Phones On 04/19/2010 02:22 PM, Jay Vocaire wrote: I have searched everywhere, but cannot seem to find anyone else talking about this issue. Maybe I am just using the wrong search terms. I am running Asterisk 1.6.2 and multiple Polycom phones all with 3.2.3 (the latest) firmware on them. I am having an issue with my 550's and my 6000's (but oddly enough, not my 320's). Whenever a number is dialed on hook, and then the speakerphone button is pressed, the number is dialed twice. If the handset is picked up, or the Dial softkey is pressed, the call is only sent once. This leads me to believe it is a phone issue, not a * config issue, but I have no way of telling. I can verify that there are two call started via the snippet below: SNIP The first hangup was triggered right away (without me doing anything), the second hangup was me actually hanging up the calling phone. It does the same thing if I dial an outside line. Any idea where to start trying to solve this? Has anyone else seen it, and can point me to the fix that I could not find with Google? Thanks. I would recommend that you enable debugging on the peer only and check to see if you see two invites come from the phone. Two invites with different call ID's would indicate it is indeed the phone making two calls. One would indicate that it MAY be an Asterisk issue. Are you using the latest Polycom firmware, btw? -- _ -- 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
Re: [asterisk-users] Odd Issue With Polycom Phones
On Wed, Apr 21, 2010 at 3:46 PM, Jay Vocaire jvoca...@innproc.com wrote: Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the tag= determines the call. The first time it sends like this: snip Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: snip The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. This is expected behavior for SIP communications. I see this all the time when an end point is registering with Asterisk. I think in those cases, however, it's a REGISTER request, not an INVITE. How is your sip.conf configured for these end points? Do you have any phones other than the ones experiencing this problem that you can test with? -- Thanks, --Warren Selby 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] Odd Issue With Polycom Phones
On 04/21/2010 03:08 PM, Warren Selby wrote: On Wed, Apr 21, 2010 at 3:46 PM, Jay Vocaire jvoca...@innproc.com mailto:jvoca...@innproc.com wrote: Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the tag= determines the call. The first time it sends like this: snip Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: snip The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. This is expected behavior for SIP communications. I see this all the time when an end point is registering with Asterisk. I think in those cases, however, it's a REGISTER request, not an INVITE. How is your sip.conf configured for these end points? Do you have any phones other than the ones experiencing this problem that you can test with? Yes this is expected behavior on a REGISTER. I didn't think that it was correct on an INVITE, however on reading RFC 3261, I believe that Asterisk is correctly responding to the request, needing credentials from the UA (Polycom). My Ekiga softphone is doing the exact same thing, however it's not creating the same 2 call issue that your Polycoms are having. The Ekiga call setup is not including credentials on the first INVITE, receives a 401 not authorized, and sends another INVITE with credentials, and receives a 100 TRYING from Asterisk. This is most likely an issue with the firmware on the Polycom. Bottom line is that another UA is doing the same thing, the call is setup properly, and it appears to work. I respectfully request that someone smarter than me take a look at this and verify my conclusions, or correct me accordingly. Thanks. According to RFC 3261 (note that the RFC uses the word request instead of register or registration request): ... If a 401 (Unauthorized) or 407 (Proxy Authentication Required) response is received, the UAC SHOULD follow the authorization procedures of Section 22.2 and Section 22.3 to retry the request with credentials. ... Read more: http://www.faqs.org/rfcs/rfc3261.html#ixzz0llyASXyI ... 22.2 User-to-User Authentication When a UAS receives a request from a UAC, the UAS MAY authenticate the originator before the request is processed. If no credentials (in the Authorization header field) are provided in the request, the UAS can challenge the originator to provide credentials by rejecting the request with a 401 (Unauthorized) status code. The WWW-Authenticate response-header field MUST be included in 401 (Unauthorized) response messages. The field value consists of at least one challenge that indicates the authentication scheme(s) and parameters applicable to the realm. An example of the WWW-Authenticate header field in a 401 challenge is: WWW-Authenticate: Digest realm=biloxi.com, qop=auth,auth-int, nonce=dcd98b7102dd2f0e8b11d0f600bfb0c093, opaque=5ccc069c403ebaf9f0171e9517f40e41 When the originating UAC receives the 401 (Unauthorized), it SHOULD, if it is able, re-originate the request with the proper credentials. The UAC may require input from the originating user before proceeding. Once authentication credentials have been supplied (either directly by the user, or discovered in an internal keyring), UAs SHOULD cache the credentials for a given value of the To header field and realm and attempt to re-use these values on the next request for that destination. UAs MAY cache credentials in any way they would like. If no credentials for a realm can be located, UACs MAY attempt to retry the request with a username of anonymous and no password (a password of ). Once credentials have been located, any UA that wishes to authenticate itself with a UAS or registrar -- usually, but not necessarily, after receiving a 401 (Unauthorized) response -- MAY do so by including an Authorization header field with the request. The Authorization field value consists of credentials containing the authentication information of the UA for the realm of the resource being requested as well as parameters required in support of authentication and replay protection. ... Read more: http://www.faqs.org/rfcs/rfc3261.html#ixzz0llyY2M2W -- _ -- 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
Re: [asterisk-users] Unable to load cdr_adaptive_odbc.so
Thanks Tilghman, this immediatley solved the problem. Perhaps a mention in cdr_adaptive_odbc.conf that the res_odbc.so module must also be loaded will help newbies like me ;) -- _ -- 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] Time difference in CSV CDR's and MySQL CDR's
Hi all, I am having a curious problem. I use two cdr backends, csv and MySQL. These are my settings: Call Detail Record (CDR) settings -- Logging:Enabled Mode: Batch Log unanswered calls: Yes * Batch Mode Settings --- Safe shutdown: Enabled Threading model:Scheduler plus separate threads Current batch size: 0 records Maximum batch size: 25 records Maximum batch time: 10 seconds Next batch processing time: 7 seconds * Registered Backends --- csv mysql cdr-custom I am finding that the calldate field varies between 3 seconds and 3 minutes between the MySQL database and the CSV files! Is this expected behaviour? I thought they should both use the same timestamp. Is is very difficult to match CDR's this way, and I am finding it hard to trust the results, as I wanted to make sure that my database was behaving correctly and not losing any CDR's along the way. Which one of the two CDR's is correct? Should this be posted as a bug? Regards, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to record a call in a single file when transfered...
On Tue, 2010-04-20 at 21:07 -0400, Leif Madsen wrote:
You could set an inherited channel variable as the first thing you do before
calling MixMonitor(). Something like:
exten = s,1,Verbose(2,Starting Call Recording) ; I always start my first
priority with something innocuous
exten = s,n,GotoIf($[${EXISTS(${CALL_RECORDED})}]?skip_rec_start)
exten = s,n,Set(__CALL_RECORDED=1)
exten = s,n,MixMonitor(${UNIQUEID}.wav,b)
exten = s,n(skip_rec_start),Verbose(2,Call recording already enabled)
exten = s,n,...
Ok. I got that working now. The only inconvenience is that I cannot
use the h extension to rename and move the wav file to its final
destination but I am getting around that by running a cron job. Thank
you very much.
Another question if I may, with variable inheritance is it possible to
do something like: Set(__CDR(userfield)=${INITIALCID})? That way I can
follow the call no matter where it is transferred to by having the
original outside callerid in the userfield.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
signature.asc
Description: This is a digitally signed message part
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to record a call in a single file when transfered...
Carlos Chavez wrote:
Another question if I may, with variable inheritance is it possible to
do something like: Set(__CDR(userfield)=${INITIALCID})? That way I can
follow the call no matter where it is transferred to by having the
original outside callerid in the userfield.
I'm not sure if it would work like that. I've never tried.
Based on my experiences with CDR's though, I don't think it'll work the way you
expect it to :)
All you can do is give it a shot and see if it works in your situation.
Leif.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.com wrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Calls drop after 20 seconds
Vieri Check out the early dial feature in the Grandstream products (if you enabled it) and play with the pedantic option. thanks, already made sure I use pedantic=no and earlydial is off in my GW Peder Like the poster below said, do a sip debug on a call and see which end sends the bye message or ends the call and go from there. That should give you some sort of clue as to who is having a timer issue. That is my next step, its just so hard to reproduce while debugging! Stefan How do you dial the users? direct with the peername or something like ex...@ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. I don't understand, all of my calls are inbound and terminated with different voip carriers, so I am not sure how that will work. I always dial d...@ipofcarrier. Will debug! Ishfaq Upgrade phones to latest/most stable firmware Upgrade routers to latest/most stable firmware This has definetly helped with other problems in the past, so I reccomend it to anybody Thank you so much for all of your help / time guys! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com wrote: Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.com wrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
On 22 Apr 2010, at 00:36, bruce bruce wrote: Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Thats.. Interesting... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
On 04/21/2010 05:36 PM, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com mailto:bruceb...@gmail.com wrote: Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.com mailto:rrb3...@gmail.com wrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want to google 'asterisk in a virtual machine' or 'asterisk timing virutal machine', or anything along those lines. I think I remember in some of the recent dahdi or asterisk release notes that they changed some settings to be more virtual machine friendly. So maybe make sure you are running the latest versions? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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 What in the world? Bruce, that is a measure of accuracy of your timing source. I believe that is the issue. What is this running on? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
It's running on an Amazon instance. No changes to system made and it was working find previously. Here is an output of top: [r...@ip-10-251-123-3 ~]# top top - 19:59:48 up 6:52, 1 user, load average: 0.78, 0.95, 0.99 Tasks: 49 total, 2 running, 47 sleeping, 0 stopped, 0 zombie Cpu(s): 0.0%us, 0.0%sy, 0.0%ni, 98.7%id, 0.0%wa, 0.0%hi, 0.0%si, 1.3%st Mem: 1740948k total, 399504k used, 1341444k free, 105300k buffers Swap: 917496k total,0k used, 917496k free, 161544k cached PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1 root 15 0 2132 752 648 S 0.0 0.0 0:00.05 init 2 root RT 0 000 S 0.0 0.0 0:00.00 migration/0 3 root 34 19 000 S 0.0 0.0 0:00.00 ksoftirqd/0 4 root RT 0 000 S 0.0 0.0 0:00.00 watchdog/0 5 root 10 -5 000 S 0.0 0.0 0:00.00 events/0 6 root 10 -5 000 S 0.0 0.0 0:00.00 khelper 7 root 11 -5 000 S 0.0 0.0 0:00.00 kthread 9 root 20 -5 000 S 0.0 0.0 0:00.00 xenwatch 10 root 10 -5 000 S 0.0 0.0 0:00.00 xenbus 17 root 20 -5 000 S 0.0 0.0 0:00.00 kblockd/0 19 root 20 -5 000 S 0.0 0.0 0:00.00 kseriod 52 root 25 0 000 S 0.0 0.0 0:00.00 pdflush 53 root 15 0 000 S 0.0 0.0 0:00.02 pdflush 54 root 20 -5 000 S 0.0 0.0 0:00.00 kswapd0 55 root 20 -5 000 S 0.0 0.0 0:00.00 aio/0 671 root 10 -5 000 S 0.0 0.0 0:00.19 kjournald 695 root 10 -5 000 S 0.0 0.0 0:00.00 kauditd 720 root 18 -4 2380 672 424 S 0.0 0.0 0:00.23 udevd 1439 root 12 -5 000 S 0.0 0.0 0:00.00 kmpathd/0 1445 root 12 -5 000 S 0.0 0.0 0:00.00 kmirrord 1463 root 10 -5 000 S 0.0 0.0 0:00.00 kjournald 1719 root 17 0 2392 572 288 S 0.0 0.0 0:00.00 dhclient 1804 root 18 0 10576 1040 752 S 0.0 0.1 0:00.34 rsyslogd 1808 root 25 0 1772 416 352 S 0.0 0.0 0:00.00 rklogd 1829 root 15 0 6948 1072 688 S 0.0 0.1 0:00.24 sshd 1858 root 25 0 2640 1208 1040 S 0.0 0.1 0:00.00 mysqld_safe 1916 mysql 15 0 118m 19m 4904 S 0.0 1.1 0:00.47 mysqld 1957 root 15 0 9480 1860 784 S 0.0 0.1 0:00.00 sendmail 1967 smmsp 18 0 8260 1488 632 S 0.0 0.1 0:00.00 sendmail 1976 root 18 0 24728 7612 4636 S 0.0 0.4 0:00.11 httpd 1992 root 18 0 3072 1128 584 S 0.0 0.1 0:00.00 crond 2005 asterisk 18 0 25476 7296 3568 S 0.0 0.4 0:00.09 httpd 2006 asterisk 15 0 25496 7300 3556 S 0.0 0.4 0:00.04 httpd 2007 asterisk 15 0 25816 7364 3596 S 0.0 0.4 0:00.11 httpd 2008 asterisk 20 0 29348 9876 4432 S 0.0 0.6 0:00.04 httpd 2009 asterisk 15 0 24888 5244 2092 S 0.0 0.3 0:00.09 httpd 2010 asterisk 17 0 25496 7300 3540 S 0.0 0.4 0:00.08 httpd 2011 asterisk 17 0 25480 7344 3572 S 0.0 0.4 0:00.07 httpd 2012 asterisk 15 0 25496 7252 3516 S 0.0 0.4 0:00.03 httpd On Wed, Apr 21, 2010 at 7:56 PM, Sean Brady sbr...@gtfservices.com wrote: On 04/21/2010 05:36 PM, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce bruceb...@gmail.com wrote: Thanks for the input. I am going to check this once I get access to system again tonight. But I thought the timing source dahdi_dummy is only good for features like MeetMe or conference rooms? or am I wrong and it has an effect on any type of calls and checking voice messages? Thanks On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock rrb3...@gmail.comwrote: So I be it sounds like all the recordings are underwater. Are you using dahdi for timing? Can you run dahdi_test? Asterisk needs a good timing source, in the case when you don't have a physical card providing it, it relies on kernel ticks or the RTC (or HPET). Because of the nature of virtual machines they don't always get access to the processor when they want and therefore their timing can get skewed and can be bad for real-time applications. There are some patches/work-arounds that you can do. You might want
Re: [asterisk-users] Asterisk choking on voice messages announcements
On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
I know that anything lower than 99% is bad. But *-400 *? Anything care of comment? Thanks, On Wed, Apr 21, 2010 at 7:45 PM, Steve Howes steve-li...@geekinter.netwrote: On 22 Apr 2010, at 00:36, bruce bruce wrote: Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Thats.. Interesting... -- _ -- 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] Asterisk choking on voice messages announcements
How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? P.S. there is only few calls load at anytime on this server. Thanks On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] ${HANGUPCAUSE} is always 0 in the h extension
Hi all,
I am using cdr_adaptive_odbc and it works fine. I am trying to save
the q931 hangupcause to a cdr record. My diaplan looks like this.
exten = _X.,1,Dial(${EXTEN})
exten = h,1,Set(CDR(q931)=${HANGUPCAUSE})
exten = h,2,Verbose(${HANGUPCAUSE})
However, as I can see by the verbose command, ${HANGUPCAUSE} is always
0. I thought it was a channel variable that contained the hangupcause?
How can I set this up to correctly save the hangupcause??
Thank you for your help
Regards,
Alex
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messagesannouncements
Bruce- How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? P.S. there is only few calls load at anytime on this server. Does Amazon cloud services guarantee you some level of performance? Do they say that if you run top you will get a 100% accurate view of compute resource usage? Your program is running on a particular server in that cloud... it would seem to me that something might be slowing that server down, and if so it's not easy to debug because you can't physically touch the server. For example it could be interesting to disconnect the server's network cable and run dahdi_test again, but you can't do that remotely. Maybe there is a simple timing measurement utility that you can run, and then show the results to Amazon, and then ask them to fix so results are acceptable. Something basic, less complex than Asterisk that Amazon tech support would agree yes if that doesn't work then it must be our problem. -Jeff On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.6.2 No soft hangup?
Steve Edwards wrote: On Tue, 20 Apr 2010, Jared Smith wrote: On Tue, 2010-04-20 at 09:49 -0700, Steve Edwards wrote: I'd like to see a more natural and intuitive interface following a verb noun model like Oracle, MySQL, or even GDB. We're close to that now, and that's one of the reasons that the soft hangup command was changed to channel request hangup. While it's not verb noun, most (if not all) of the commands in the Asterisk CLI should follow the module verb noun model. Having to know which module implements a command is an obstacle. Being the 1.2 Luddite that I am, I'll withhold further criticism until I play with 1.6 a bit. An apropos command would be a nice addition. This is a great idea. At least help hangup should show any commands that include hangup and their description. Now: asterisk*CLI help hangup hangup request no description available So you need to know what's the answer before you can see it: asterisk*CLI help channel request hangup Usage: channel request hangup channel Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel sean -- _ -- 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] Security tests
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! In the network of my house I was testing the security with my Asterisk installation. The first test that I'm doing is an man in the middle attack. In this scenary, the attacker is a virtual machine that it tries to see the SIP traffic between a PC with a softphone and a Grandstream BT200 telephone. But it draws attention to me between the PC with softphone and the telephone I see traffic ARP or ICMP that could make to try between the equipment but does not see RTP. Is there some special consideration that it must to observe? I am doing it to the capture with: # tcpdump -i eth0 -n host 10.1.0.65 -w dump where 10.1.0.65 is the PC with softphone. Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkvPpYAACgkQZpa/GxTmHTenpwCfcL3gBTTf0jRiEpv0k+jf2GkP WR8An2RxSdFdkdyRntOmVUof5kOygLYB =EG9x -END PGP SIGNATURE- -- _ -- 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] Improving CLI Help - was [Re: 1.6.2 No soft hangup?]
Olle E. Johansson wrote: Further to Steve Edward's comment, I think things would be more obvious if the help system was improved slightly, for instance: If you were trying to figure out the commands dealing with peers, you would be able to type: *CLI help peer No peer command found. Possible alternatives: iax2 show peer Show details on specific IAX peer iax2 show peers List defined IAX peers sip show peers List defined SIP peers sip show peer Show details on specific SIP peer (and so on, maybe using the [More] option to help it be readable) In this case, if I could use the help system to search on all occurrences of the word hangup in the available commands, I would probably have found it myself instead of bothering the list. THat's a very good idea. Thank you! Now we need someone that codes it :-) /O Well I'm certainly not the one who could code it, but is there any way to simply grep all the help. So, for instance, if you did help hangup you got: hangup request no description available which you now get, followed by all the commands that have hangup in them, including their descriptions. For instance: But see: channel request hangup channel Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel etc etc sean -- _ -- 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] Improving CLI Help - was [Re: 1.6.2 No soft hangup?]
Hi,
Maybe you could do something in shellscript too:
e.g.
asterisk -rx help | grep -ia something
That would behave just as describe in the suggestion (but it's easier to
do :P)
You could place that in a tiny shellscript, that takes the 'something'
as an argument:
#!/bin/bash
token=$1
asterisk -rx help | grep -ia ${token}
Save that with the name of your preference (somewhere inside your $PATH,
would be nice), and just execute it like a normal command:
[name_you_gave_it] sip
[name_you_gave_it] peer
[name_you_gave_it] whatever
Note:
You then could make a lot of fancy customizations to parameters of your
script, etc., and even use other tools for if needed (e.g. gawk, sed, etc.)
sean darcy wrote:
Olle E. Johansson wrote:
Further to Steve Edward's comment, I think things would be more
obvious if the help system was improved slightly, for instance:
If you were trying to figure out the commands dealing with peers, you
would be able to type:
*CLI help peer
No peer command found. Possible alternatives:
iax2 show peer Show details on specific IAX peer
iax2 show peers List defined IAX peers
sip show peers List defined SIP peers
sip show peer Show details on specific SIP peer
(and so on, maybe using the [More] option to help it be readable)
In this case, if I could use the help system to search on all
occurrences of the word hangup in the available commands, I would
probably have found it myself instead of bothering the list.
THat's a very good idea. Thank you!
Now we need someone that codes it :-)
/O
Well I'm certainly not the one who could code it, but is there any way
to simply grep all the help. So, for instance, if you did help
hangup you got:
hangup request no description available
which you now get,
followed by all the commands that have hangup in them, including their
descriptions. For instance:
But see:
channel request hangup channel
Request that a channel be hung up. The hangup takes effect
the next time the driver reads or writes from the channel
etc
etc
sean
--
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs
--
_
-- 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] ${HANGUPCAUSE} is always 0 in the h extension
However, as I can see by the verbose command, ${HANGUPCAUSE} is always
0. I thought it was a channel variable that contained the hangupcause?
Just an update, if the call is established, then there is a
hangupcause received.
The above problem only happens if the caller hangs up before pickup.
This is usualy a cause 16, not 0.
Alex
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk choking on voice messages announcements
On Wed, 21 Apr 2010, bruce bruce wrote: It's running on an Amazon instance. No changes to system made and it was working find previously. Maybe you could correlate the fluctuations in your timing source with the attacks on Randy and Fred's systems. -- 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] Asterisk choking on voice messages announcements
On Wed, Apr 21, 2010 at 6:13 PM, bruce bruce bruceb...@gmail.com wrote: How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? P.S. there is only few calls load at anytime on this server. Here are few ideas: 1. I have seen complaints that as Amazon loads up its virtual machines, that neighboring VM's running on the same hardware are sucking up CPU cycles and reducing the performance of the other VM's on board. One guy was complaining that to get the same performance he got a few months ago, he has to move to a more powerful machine, which costs more . You might move up to a more expensive, faster VM and see if it helps. 2. I don't know exactly how Dahdi gets its timing, but I do know that it has two methods; one involves HIGH RES TIMERS compiled into the kernel. The other when the high-res stuff isn't included. You can decompress /proc/config.gz into a local file and look for HIGHRES to be defined. If it isn't you might try to find a kernel with it defined, and see if it helps. 3. If you are on 1.6.1 or 1.6.2 (too tired to look up which), you could try using another method of generating timing than dahdi_dummy. I suspect that they may just reflect code already in Dahdi_dummy... but this seems like something you might want to become knowledgeable about! murf Thanks On Wed, Apr 21, 2010 at 8:03 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Wed, 2010-04-21 at 19:36 -0400, bruce bruce wrote: Here are result of dahdi_test: [r...@ip-10-251-123-3 ~]# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434% -434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770% 99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622% 98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478% 94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797% 98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921% 91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212% What can one tell from these? Only that your timing source sucks. You need 99.9% or higher if you want a stable system. I have servers with dahdi_dummy that never go below 99.7% accuracy. You really need to check your timing source. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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 -- Steve Murphy ParseTree Corp -- _ -- 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] Calls drop after 20 seconds
Alejandro Recarey schrieb: Stefan How do you dial the users? direct with the peername or something like ex...@ipofpeer ? i know this problem when dialing a patton ISDN ata without an extension. The call is established but when the T1 sip timeout fires the call gets disconnected. Maybe you could do some sip debugging and watch for resend sip messages. I don't understand, all of my calls are inbound and terminated with different voip carriers, so I am not sure how that will work. I always dial d...@ipofcarrier. Will debug! what i mean is that the problem what i have was when i dial no exten directly via the IP of the patton. which looks like this: Dial(SIP/@123.123.123.123,120) when this happens the T1 Timeout ends the call after 30 seconds. this only happens on inbound calls to the customers, not outbound to a carrier. best regards. steve -- _ -- 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
