[asterisk-users] Audio Problems - Operating System??
Hey All, I've been using Asterisk for a couple years now, but have always had some unsolvable audio problems. I get audio stuttering and popping quite often. Even if I have just one call up! The server is a Dual Duo-Core 3.0ghz Xeon Processor PC with 4 GB or ram. It just seems to me that this should NOT be happening. The server resources are nearly 98% idle. I've tried using the SLN audio file format, which does reduce the CPU usage when playing audio files, but it didn't help the audio quality. I've also tried putting my audio files on a RAM Drive and still have the same problem. I've also slimmed my asterisk system down to load only the modules that I am using via modules.conf. Now my question. I've heard through the grapevine that the Operating system running Asterisk can make a big difference in performance. I am currently running SuSE Linux Enterprise Server 10.A friend of mine actually talked to someone at Digium about this specific problem and they told him -not- to run SuSE. Is this correct? Has anyone else had any experience similar to this? I'm just wondering if Digium just wanted to push Asterisk Business Edition running on rPath on him, or if there really are some conflicts with SuSE that may cause audio instability. If so then it definitely would explain a lot regarding my poor audio quality problems. I would be happy to hear thoughts that any of you might have. Thanks so much! Darren Nay [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with rxfax
Just out of curiosity. Would you mind sharing that app_rxfax.c file that you modified to work with SpanDSP 0.0.3? TIA, Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ardjan Zwartjes Sent: Monday, January 22, 2007 2:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems with rxfax Dear list, The company I'm working for is trying to use app_rxfax to receive faxes on an Asterisk machine. Our initial tests looked very promising, but unfortunately we've encountered some problems. We've been trying to solve these problems for quite some time now, but we're running out of options. So I really hope that somebody can give some help here. Basically our set-up is this: We have an Asterisk server (version 1.2.7) with an ISDN trunk (Sangoma A104D), we've configured asterisk to run rxfax on a specified extension. Originally we started out with spandsp 0.0.2pre26 and the original app_rxfax for spandsp 0.0.2. Some of the faxes were coming in perfectly, but soon we noticed that quite often there were substantial pieces of the fax missing in the resulting tif file. We've tried the following to solve these problems: - We've checked the timing settings for the ISDN trunk, these seem to be ok. - We've tried several versions of libtiff (currently we are using 3.7.2). - We've tried using 0.0.3 versions of spandsp (since we're using asterisk 1.2.7 we had to modify app_rxfax.c to work). - We've created a custom dialplan application to disable the echo cancellation on the isdn channel on which the fax is received. - We've tried various settings for t30_set_supported_compressions, t30_set_supported_image_sizes, t30_set_supported_modems and t30_set_supported_resolutions (I must confes that I didn't really know which settings to use here, but we have tried a lot of them). - We've tried several fax machines to send the faxes, ranging from simple fax-modems to large dedicated fax machines. But still a lot of faxes give problems, either the tif is missing large portions, or the fax isn't received at all. At the bottom of this mail are 2 examples of the logging when it goes wrong. I really hope that somebody can give a few pointers. Thanks in advance, Kind regards, Ardjan Zwartjes, Telecats === Example 1 = Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:14 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:17 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:19 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:20 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-22.18dBm0) Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:21 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:23 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-17.84dBm0) Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:38 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:39 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 8 Jan 19 14:49:41 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:42 DEBUG[24218] app_rxfax.c: FLOW FAX Switching from V.29 + V.21 to V.29 (-19.13dBm0) Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 4 Jan 19 14:49:48 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Pages transferred: 1 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size: 1728 x 1192 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image resolution8037 x 7700 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Transfer Rate: 9600 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Bad rows0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Longest bad row run 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Compression type3 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: Image size (bytes) 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: == Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:49 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 4 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set rx type 0 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: FLOW FAX Set tx type 1 Jan 19 14:49:51 DEBUG[24218] app_rxfax.c: ==
[asterisk-users] Streaming audio file while working in background ?
Hey All, Is there an app available, or another method, to stream an audio file to a caller while performing additional actions in the background? Regardless of whether DTMF is received or not from the caller. I had originally thought that I could use the Background app for this but after further investigation found that Background is primarily for playing audio and waiting for DTMF, and it seems it won't do what I need in this situation. Ideally I would like to be able to play an audio file to the caller while making outbound calls in the background (via the Dial app) and then discontinue the audio file stream and bridge the calls once an outbound call is connected. Can anyone point me in the right direction on how to do this with * ? Thanks in advance, Darren Nay ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cisco 7970 Unprovisioned
Sounds like you need to dig into the documentation for the 7970 and perhaps even contact Cisco TAC if that doesn't help. It doesn't sound like your problem is related to Asterisk. The Cisco IP phone won't register with asterisk until it's been provisioned. Those 7900 series cisco phones are very finicky. Best of luck! From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Token PBX Sent: Saturday, January 20, 2007 6:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Cisco 7970 Unprovisioned Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written Unprovisioned, and phone is not trying to register with asterisk. Please help!! Mihaela ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, SpanDSP and RXFax
Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax application is invoked: Jan 16 09:29:11 NOTICE[5414]: channel.c:1950 ast_read: Dropping incompatible voice frame on SIP/192.168.2.250-b3203b30 of format slin since our native format has changed to ulaw Strange thing is that after that error Asterisk will sit and wait for the FAX to complete: Executing Application: (rxfax) Options: (/tmp/11689649465416/fax.tiff) When the fax is completed then the sending fax machine always says that the fax was sent successfully, but Asterisk errors out of the rxfax application and never writes the fax.tiff file. Has anyone seen this behavior before? Any help that you could provide would be very much appreciated. Thanks in advance! Darren Nay [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Asterisk, SpanDSP and RXFax
Hey All, Nevermind this question. I figured out that my problem was that I needed to downgrade my libtiff library to v3.7.1. My OS had installed 3.8.2 during system install and apparently spandsp doesn't like that version. It's all working perfectly now. Thanks in any case! Darren Nay From: Darren Nay Sent: Tuesday, January 16, 2007 10:25 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk, SpanDSP and RXFax Hey All, I am attempting to get the RXFax app working and having a hell of a time of it. I am hoping that some of you fine folks can help me out. I have installed Asterisk v1.2.14, SpanDSP v0.0.2pre26 and app_rxfax. All compiled and installed fine. When I attempt to call the extension I have created for receiving fax's then I get the following error once just as the rxfax application is invoked: Jan 16 09:29:11 NOTICE[5414]: channel.c:1950 ast_read: Dropping incompatible voice frame on SIP/192.168.2.250-b3203b30 of format slin since our native format has changed to ulaw Strange thing is that after that error Asterisk will sit and wait for the FAX to complete: Executing Application: (rxfax) Options: (/tmp/11689649465416/fax.tiff) When the fax is completed then the sending fax machine always says that the fax was sent successfully, but Asterisk errors out of the rxfax application and never writes the fax.tiff file. Has anyone seen this behavior before? Any help that you could provide would be very much appreciated. Thanks in advance! Darren Nay [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remote-Party-ID + CallerID + VoicemailMain
Hey All,
Quick Question. We just started using Remote-Party-ID
on our IAD endpoints and now when one of our customers has caller-ID blocked
(Privacy=full in the remote-party-id SIP header) and they call voicemail via
asterisks and get VoiceMailMain then they get a prompt for "Comedian
Mail, Mailbox?" instead of just the password.
We are calling VoiceMailMain as: VoiceMailMain(${CALLERIDNUM})
So, after some investigation it seems that the reason is
because the CALLERIDNUM and CALLERID variables now always contain a value of "Anonymous"
when the "Privacy" flag is set in the Remote-Party-ID SIP header.
Is there a way to disable Remote-Party-ID in Asterisk?
So that asterisk always looks at the SIP From: header instead of
Remote-Party-ID?
Or, is there a variable that I am unaware of that contains
the calling-number other than caller-id? I just need the calling number
available somewhere ... I can easily use an AGI script to parse it out of a
string and pass it to VoiceMailMain .. I just need access to it from an AGI
script in order to do that.
Any ideas?
Thanks for any help you can provide!
Darren Nay
Ionosphere, Inc.
VoIP Network Development
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY re quests)
Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay -Original Message- From: John Todd [mailto:[EMAIL PROTECTED] Sent: Saturday, May 22, 2004 1:57 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Sipura and STUN (was: rejected NOTIFY requests) At 7:36 PM +0200 on 5/22/04, Olle E. Johansson wrote: [snip] Sending NOTIFY to Asterisk is an error, but a workaround. Since Asterisk can handled the NAT traversal all by itself with Qualify (as John points out) disabling the NOTIFY will not change anything. The NOTIFY will in no way affect the status - unreachable/reachable. Another problem with the SIPURA is the lack of a working STUN solution. Even Grandstream works better with NAT today. /O Do you have difficulties with the Sipura SPA-2000 (or other Sipura boxes) and Asterisk? I've found no problems, even behind NAT, though I have only tried behind one or two NAT devices (OpenBSD and Apple Airport.) It's surprising that Sipura doesn't include STUN as an option - their list of options is so huge that I always assumed I had just missed it, but now that I look closer I suppose you're right. Do Asterisk users even really need STUN? I've never found it to be required after the NAT issues were worked out of Asterisk... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail storage in DB
Hey all, Quick Question. I have heard mention that Asterisk has the capability to store voicemail inside a database, instead of storing each voicemail in a separate file under a spool directory. Is this true? If so, does it (or can it) use MySQL? Is there any documentation available showing how to do this? The problem that we are having is that we need redundant voicemail servers and in order to do that we would need to replicate the voicemail "spool" directory to each redundant server ... we haven't been able to find an efficient -yet- cost effective method for this. However, if we can use a mysql database for voicemail storage then I can set up mysql database replication and our problem is solved. Thanks so much for your help! Darren Nay - [EMAIL PROTECTED]
[Asterisk-Users] Newbie question
Hey All, We are using Asterisks as a voicemail only application, and so far all is great. (Excellent product!) However, I do have one question that I am hoping you might be able to help me with. In our asterisk application. When our customers call *55 (our dialplan code to check voicemail) then they are sent directly to voicemail (asterisk). Asterisk then gives a voice prompt asking the customer to enter their extension number (entire 10 digit telephone number in our case). My question is. Is there a way to make asterisk aware of the calling-from (callerID) number so that it will automatically detect the number and then go directly to asking them to input their password. If so, where would I make the config changes for this in the asterisk config files, and does anyone have an example of a similar config? Thanks! Darren Nay VOIP Network Developer Ionosphere, Inc [EMAIL PROTECTED]
[Asterisk-Users] MySQL Dynamic Extensions
Hello All, I am just looking into Asterisk as a viable voicemail solution for our phone service. In order to use it though I will need to make extensions.conf dynamic (ie. Via MySQL). Is this possible? I've found the following information on this subject: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql However, this is not a fully dynamic function. It requires me to pull the mysql database every so often (presumably via cron) and then restart asterisk after updating extensions.conf. Is it possible to setup asterisks so that extensions.conf is fully dynamic via a MySQL database? Thanks for the help!! Regards, Darren Nay [EMAIL PROTECTED]
