[asterisk-users] Queue calls to agent end prematurely with diastatus cancel
Previously posted to the Users list (FYI). We have a system running Asterisk-1.4.40. Queue calls are distributed using rrmemory with a 20 second timeout. What we are seeing is; when a call in the queue will call the first agent for 20 seconds, and subsequent attempts will call agents for random periods of time (as little as one second), and continue on to the same or next agent. When this happens, the dialstatus variable is set to Cancel. This suggests that the queue is canceling the call to the agent, but we can find no configuration or error logging to show why this is happening. I also was unable to find any bugs logged on this issue. How can we further troubleshoot this issue? Chris queues.conf [myqueue] strategy = rrmemory joinempty = strict leavewhenempty = strict ringinuse=no monitor-join=yes monitor-format=wav monitor-type = MixMonitor context=ss-queueout servicelevel = 180 wrapuptime = 0 timeout = 20 retry = 0 weight = 0 -- _ -- 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] Queue calls to agent end prematurely with diastatus cancel
We have a system running Asterisk-1.4.40. Queue calls are distributed using rrmemory with a 20 second timeout. What we are seeing is; when a call in the queue will call the first agent for 20 seconds, and subsequent attempts will call agents for random periods of time (as little as one second), and continue on to the same or next agent. When this happens, the dialstatus variable is set to Cancel. This suggests that the queue is canceling the call to the agent, but we can find no configuration or error logging to show why this is happening. I also was unable to find any bugs logged on this issue. How can we further troubleshoot this issue? Chris queues.conf [myqueue] strategy = rrmemory joinempty = strict leavewhenempty = strict ringinuse=no monitor-join=yes monitor-format=wav monitor-type = MixMonitor context=ss-queueout servicelevel = 180 wrapuptime = 0 timeout = 20 retry = 0 weight = 0 -- _ -- 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] Queue calls to agent end prematurely with diastatus cancel
We have a system running Asterisk-1.4.40. Queue calls are distributed using rrmemory with a 20 second timeout. What we are seeing is; when a call in the queue will call the first agent for 20 seconds, and subsequent attempts will call agents for random periods of time (as little as one second), and continue on to the same or next agent. When this happens, the dialstatus variable is set to Cancel. This suggests that the queue is canceling the call to the agent, but we can find no configuration or error logging to show why this is happening. I also was unable to find any bugs logged on this issue. How can we further troubleshoot this issue? Chris queues.conf [myqueue] strategy = rrmemory joinempty = strict leavewhenempty = strict ringinuse=no monitor-join=yes monitor-format=wav monitor-type = MixMonitor context=ss-queueout servicelevel = 180 wrapuptime = 0 timeout = 20 retry = 0 weight = 0 -- _ -- 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] T.38 Peer Negotiation Fails
Asterisk 1.4.32 (Also 1.4.26, 1.4.33) Broadvox ITSP (xxx.xxx.xxx.xxx) Linksys 2102(yyy.yyy.yyy.yyy) Both peers : canreinvite=yes t38pt_udptl = yes I'm having some trouble getting a T.38 fax call established with Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38 switchover) to Broadvox with the Asterisk server's IP address in the Connection Information (c) instead of the Linksys ATA's IP address. This causes the negotiation to revert back to t38state zero (chan_sip.c: T38_DISABLED), and shortly after the ATA hangs up. What is a bit odd about this, is that Asterisk says it's about to establish a peer to peer UDPTL connection : chan_sip.c: Sending reinvite on SIP '1057817983_43059...@xxx.xxx.xxx.xxx' - It's UDPTL soon redirected to IP yyy.yyy.yyy.yyy:16468 chan_sip.c: Strict routing enforced for session 1057817983_43059...@xxx.xxx.xxx.xxx On a known good/working T.38 configured Asterisk PBX elsewhere (with Affinity as the ITSP), I also see the Strict routing message, yet T.38 negotiation achieves t38state 5 (chan_sip.c: T38_ENABLED) and calls are successful. I've been comparing Asterisk debug from both systems as well as wireshark captures, but I can't figure out why Asterisk is not sending the Linksys ATA's IP address. Broadvox uses a Sonus switch and gateway with separate IP addresses for SIP and media. Affinity uses Sippy (?) with a common IP for SIP and media. I believe I've already covered all the possible configuration scenarios. I just can't get the right detail out of Asterisk to determine if this is an Asterisk issue, or an ITSP issue. Using bug ID#16705 as a guide, I patched this version, as well as downgraded to a known working version, and to the latest 1.4.33.1 which includes several t.38 fixes. https://issues.asterisk.org/view.php?id=16705 Thoughts? Chris -- _ -- 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-gplonly dependency in asterisk-addons RPM
It seems that asterisk-addons and one or more of Digium's licensed modules such as res_fax_digium have a conflict that doesn't seem to be documented anywhere I can find. In a nutshell, asterisk14-addons-core has a fake provide for asterisk-gplonly : # # core subpackage # %package core Summary: Asterisk-addons core package. Group: Utilities/System Provides: asterisk-gplonly Provides: asterisk-addons-core Obsoletes: asterisk-addons-core Requires: asterisk14-core The Digium licensed packages look for this package and prevent installation : --- Package asterisk14-res_fax.i386 1:1.4_1.0.14-1_centos5 set to be updated --- Package asterisk14-res_fax_digium.i386 1:1.4_1.0.11-1_centos5 set to be updated -- Processing Conflict: asterisk14-res_fax conflicts asterisk-gplonly -- Processing Conflict: asterisk14-res_fax_digium conflicts asterisk-gplonly -- Finished Dependency Resolution 1:asterisk14-res_fax_digium-1.4_1.0.11-1_centos5.i386 from digium-current has depsolving problems -- asterisk14-res_fax_digium conflicts with asterisk14-addons-core 1:asterisk14-res_fax-1.4_1.0.14-1_centos5.i386 from digium-current has depsolving problems -- asterisk14-res_fax conflicts with asterisk14-addons-core Error: asterisk14-res_fax conflicts with asterisk14-addons-core Error: asterisk14-res_fax_digium conflicts with asterisk14-addons-core A comment in the spec file would have been nice... Does anyone know if this a real technical issue, or simply a licensing conflict between GPL and Digium? Chris -- _ -- 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-gplonly dependency in asterisk-addons RPM
On 4/1/2010 1:52 PM, Kevin P. Fleming wrote: Chris Miller wrote: A comment in the spec file would have been nice... Does anyone know if this a real technical issue, or simply a licensing conflict between GPL and Digium? It is not a technical issue; it is an issue because some of the modules in -addons have licenses that are pure GPLv2 only, and in addition the license for MySQL-based components restricts their usage to *only* GPLv2-licensed applications unless a commercial license for MySQL is obtained. Since loading one of Digium's binary modules into an Asterisk process changes it to no longer be pure GPLv2, such usage restrictions should be taken into account. The purpose of that conflict is to ensure that the person installing the packages is made aware of the issue and that they must take explicit action to override it (thus ensuring that we don't facilitate accidental violation of third-party license agreements). Understood, I figured it was something like that. Do you have some mechanism in the source install that causes similar enforcement behavior? If you can suggest a method to provide this information to people in some automatic way when they are made aware of the conflict by RPM, feel free to do so and we'll try to get it incorporated into the RPMs themselves. A method of providing the GPL license conflict information at install time, or the reason for (and resolution of) the RPM install conflict? It seems to me that the GPL information could be displayed in the register binary since no end user can use a Digium supplied commercial module without registration, right? It could also be displayed on the Digium website where end users have to purchase their Digium licenses. This begs the question of when the actual violation occurs. In other words, is this really a usage issue, or does the violation occur at install time even though the non-GPL component is not usable? It sounds to me that many users are violating the GPL by installing the non-GPL modules. Rather than simply making it difficult to install, why not be proactive in encouraging compliance by detailing the steps openly. When I Googled for this issue, I turned up no useful information. Seems like a page explaining the above somewhere on the Asterisk and/or Digium site would be helpful. The only workaround at this point is to force install the RPMs. This encourages lesser skilled sysadmins to use this practice regularly (on all Linux dependency issues) without fully understanding what they are doing. I took the time to download the SRPM and saw this was an arbitrary dependency, but most sysadmins won't burn the time. What also concerned me was a few posts about a system stability issue with the SkypeForAsterisk module after force installing the RPM. This contributed to my being uneasy about proceeding with this route without full knowledge of the situation. Alternatively I need to maintain my own version of asterisk-addons-core without the gplonly provide. Kinda defeats the purpose of using a third party repository for convenience. I understand the reasons why this was done, but unless I've overlooked some resource on the interwebs, it looks like the other shoe never dropped and zero documentation was provided to work with this issue. I can't think of a clean way off the top of my head to address this in RPM, so I'd argue that RPM is simply not the appropriate choke point to enforce compliance. Feel free to send me a PM if you want to discuss further. Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. (831) 621-7928 http://www.scratchspace.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] Restarting of B-channel on span 1
Darrin Henshaw wrote:
add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
depending on what you are running. zaptel or dahdi.
Can someone confirm when the default was changed from never to
3600 seconds? According to the voip-info wiki, never has always
been the default. I would tend to agree, because I've never seen
this behavior on customer systems until recently.
http://www.voip-info.org/wiki/view/Asterisk+config+chan_dahdi.conf
The current configs/chan_dahdi.conf.sample says the default is 3600.
I don't see anything in the 1.4 changelog about this change.
I ran across this troubleshooting a PRI issue, and was concerned
that this frequent resetting was related to the customer issues.
What happens when a call comes in when a reset is in progress? If
this condition can't be handled gracefully (i.e. without failing a
call), then I would argue the default is not conservative enough.
Just want to know the right way to handle this.
Chris
On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating after every interval of one hour, and i have search
and psot on
fourms and everyone said that this is a normal behaviour.
If this is a normal behaviour is there is any way to stop it { i still don't
know what is the reson to restart ever hour } . Because this is listed
everywhere that this is a normal behaviour, but not one mention {may be i am
not able to find it is listed some where} why this is nesessary? and if this
is not nessary how to stop it...
I think we all already know the message , but posting it for future
reference..
Thanks a lot .
Aman Dhally
--
ul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Event Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: Asterisk Queue Logger restarted
[Jul 8 04:02:03] VERBOSE[9007] logger.c: -- Remote UNIX connection
disconnected
[Jul 8 04:51:30] VERBOSE[3300] logger.c: -- B-channel 0/1 successfully
restarted on span 1
[Jul 8 04:51:35] VERBOSE[3300] logger.c: -- B-channel 0/2 successfully
restarted on span 1
[Jul 8 04:51:40] VERBOSE[3300] logger.c: -- B-channel 0/3 successfully
restarted on span 1
[Jul 8 04:51:45] VERBOSE[3300] logger.c: -- B-channel 0/4 successfully
restarted on span 1
[Jul 8 04:51:50] VERBOSE[3300] logger.c: -- B-channel 0/5 successfully
restarted on span 1
[Jul 8 04:51:55] VERBOSE[3300] logger.c: -- B-channel 0/6 successfully
restarted on span 1
[Jul 8 04:52:00] VERBOSE[3300] logger.c: -- B-channel 0/7 successfully
restarted on span 1
[Jul 8 04:52:05] VERBOSE[3300] logger.c: -- B-channel 0/8 successfully
restarted on span 1
[Jul 8 04:52:10] VERBOSE[3300] logger.c: -- B-channel 0/9 successfully
restarted on span 1
[Jul 8 04:52:15] VERBOSE[3300] logger.c: -- B-channel 0/10 successfully
restarted on span 1
[Jul 8 04:52:20] VERBOSE[3300] logger.c: -- B-channel 0/11 successfully
restarted on span 1
[Jul 8 04:52:25] VERBOSE[3300] logger.c: -- B-channel 0/12 successfully
restarted on span 1
[Jul 8 04:52:30] VERBOSE[3300] logger.c: -- B-channel 0/13 successfully
restarted on span 1
[Jul 8 04:52:35] VERBOSE[3300] logger.c: -- B-channel 0/14 successfully
restarted on span 1
[Jul 8 04:52:40] VERBOSE[3300] logger.c: -- B-channel 0/15 successfully
restarted on span 1
[Jul 8 04:52:45] VERBOSE[3300] logger.c: -- B-channel 0/17 successfully
restarted on span 1
[Jul 8 04:52:50] VERBOSE[3300] logger.c: -- B-channel 0/18 successfully
restarted on span 1
[Jul 8 04:52:55] VERBOSE[3300] logger.c: -- B-channel 0/19 successfully
restarted on span 1
[Jul 8 04:53:00] VERBOSE[3300] logger.c: -- B-channel 0/20 successfully
restarted on span 1
[Jul 8 04:53:05] VERBOSE[3300] logger.c: -- B-channel 0/21 successfully
restarted on span 1
[Jul 8 04:53:10] VERBOSE[3300] logger.c: -- B-channel 0/22 successfully
restarted on span 1
[Jul 8 04:53:15] VERBOSE[3300] logger.c: -- B-channel 0/23 successfully
restarted on span 1
[Jul 8 04:53:20] VERBOSE[3300] logger.c: -- B-channel 0/24 successfully
restarted on span 1
[Jul 8 04:53:25] VERBOSE[3300] logger.c: -- B-channel 0/25 successfully
restarted on span 1
[Jul 8 04:53:30] VERBOSE[3300] logger.c: -- B-channel 0/26 successfully
restarted on span 1
[Jul 8 04:53:35] VERBOSE[3300] logger.c: -- B-channel 0/27 successfully
restarted on span 1
[Jul 8 04:53:40] VERBOSE[3300] logger.c: -- B-channel 0/28 successfully
restarted on span 1
[Jul 8 04:53:45] VERBOSE[3300] logger.c: -- B-channel 0/29 successfully
restarted on span 1
[Jul 8 04:53:50] VERBOSE[3300] logger.c: -- B-channel 0/30 successfully
restarted on span 1
[Jul 8 04:53:55] VERBOSE[3300] logger.c: -- B-channel 0/31 successfully
restarted on span 1
[asterisk-users] Zaptel ring voltage detection
We've inherited a pair of mostly identical PBX systems, each with a TDM400P Rev I boards and 4 FXO modules. The production system is running Asterisk-Now with 1.4.9, and despite some other issues, it is able to answer inbound calls just fine. The replacement system is currently running Asterisk 1.2.28, and is unable to detect incoming calls, outbound calls work fine. We discovered later that the analog lines are supplied by Cox Cable in Los Angeles, apparently Cox is the only telco available in this office building. I came to the conclusion that Cox is probably providing analog lines via Cable/VOIP service and that the FXS ports in their equipment are providing a lower than normal ring voltage. ztmonitor shows the ring, but Asterisk 1.2 never starts a simple switch on the zap channel. I did find a posting from the 1.0.X days where someone had to lower the sample peaks in wcfxo.c from +-32000 to +-1, but this seems to be the standard in the current 1.2 zaptel source (wctdm.c of course). Clearly the 1.4 zaptel driver is doing something different, but I'm not sure what since the values look the same. We've been avoiding Asterisk 1.4 because of some serious stability issues we've seen in several versions. 1.4.19 seems to have addressed these issues, so we're likely going to deploy the replacement PBX with 1.4.19. Just mentioning it because I'm sure someone will ask ;-). While I hope to never run into this issue again, it does raise the question of how could one determine the actual ring voltage, as well as any other analog line values that would help troubleshoot this sort of issue? If there are further issues with the analog lines, we may need the ability to detect and tweak the ring detection parameters. Thoughts? Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VPM450: Not Present
I've got a system with a TE412P installed under Fedora Core 6 and I continue to see this message in the logs. The card most certainly does have an EC module installed. The system is suffering from echo problems, and I suspect this is no coincidence... I've double checked to ensure the module has been inserted correctly. I've not seen any other complaints on the lists, etc. about this error message, so I'm running out of clues. Same problem under Fedora Core 4. How does one confirm/troubleshoot EC card detection? Chris___ --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] Audio playback problems with FC6 and Zaptel 1.2.16
I'm chasing down some issues at a call center. Today I received a complaint that audio file playback ceased after they upgraded the system from FC4 to FC6, Asterisk 1.2.14 to 1.2.17. Zaptel is at 1.2.16. The system in question takes inbound calls via IAX2 and has a TE410P with a couple of channel banks connected to it for analog extensions. I ultimately found that the problem goes away if I load ztdummy alone or prior to wct4xxp. I realize ztdummy should not be used when there's real hardware available, but it appears to solve/mask the problem at least for troubleshooting. No errors or clues in the logs, dmesg, etc. I even tried transcoding the gsm audio files in ulaw with no luck. As an aside, I noticed that zttranscode loads itself when Asterisk is started. I haven't found anything in Mantis, Google, etc. Before I file a bug report, I wanted to see if anyone else has seen this weirdness. Chris ___ --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] Handling SIP 482 condition
Eric ManxPower Wieling wrote: Chris Miller wrote: I would tend to agree, but the context that holds these number is an inbound context which includes additional logic that would fail normal calls. Yes, I can add the DIDs to the outbound context, but the point here is not to have a bloated dialplan with parallel data in multiple contexts. If I must have parallel data, I'd rather do a lookup in an external table using AstDB or an application similar to DUNDILookup() or ENUMLookup(). Another route I tried was to setup a local SIP trunk to catch the loops and send them down the inbound context. This fails because there are no SIP headers and the unknown peer is effectively NULL and will never match this trunk. As I said, they just get routed to from-sip-external. Put the DIDs in a context by themselves. include = that context in both your incoming context and your phones context. Thanks for the reply. I know this will work and am already doing this as a temporary workaround, but this doesn't really scale with hundreds/thousands of DIDs. I'm trying to avoid a bloated dialplan and the DIDs are already listed in another context, taking up space. What I'm looking for is some way to catch 482 loops and treat them as inbound calls without resorting to a parallel context. Failing that, I'd like to perform efficient lookups in an external DB, perhaps killing two birds with one stone (all DIDs can just exist in the DB). Chris ___ --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] Handling SIP 482 condition
Asterisk SVN-branch-1.2-r48484 I get a SIP Response 482 (loop detected) back from my SIP provider whenever I dial from/to DIDs on the same server. The call is assumed from an unknown peer, then gets routed to Local/DID@from-sip-external which fails. No SIP headers/messages are generated because the SIP channel is gone. It all makes sense, but how can I go about telling Asterisk not to dial out of a trunk when the number is local? I could list the DIDs under from-sip-external, but that would potentially allow anyone to connect to the server by spoofing the DID. Seems like there ought to be an easy way get Asterisk to consult it's own inbound DID routes before selecting an outbound trunk, and without populating the dialplan with a parallel list of DIDs. I can't imagine I'm the only one to have run into this, but there's nothing on the lists about this scenario. Chris ___ --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] Handling SIP 482 condition
Paul Hales wrote: But the general thought is that if you build your contexts right, your internal SIP users should hit those numbers as part of their dialplan. I would tend to agree, but the context that holds these number is an inbound context which includes additional logic that would fail normal calls. Yes, I can add the DIDs to the outbound context, but the point here is not to have a bloated dialplan with parallel data in multiple contexts. If I must have parallel data, I'd rather do a lookup in an external table using AstDB or an application similar to DUNDILookup() or ENUMLookup(). Another route I tried was to setup a local SIP trunk to catch the loops and send them down the inbound context. This fails because there are no SIP headers and the unknown peer is effectively NULL and will never match this trunk. As I said, they just get routed to from-sip-external. The following BUG describes this issue, however the suggested patch doesn't appear to have made it into the stable code. http://bugs.digium.com/view.php?id=7403 I can't apply the patch due to massive changes in chan_sip.c since the revision (47646) the patch was designed for, as a result I don't feel comfortable trying to hack the patch code into the current version. Just trying to avoid reinventing the wheel if there's already a known workaround. Chris ___ --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] channel.c: Nobody there, continuing...
I'm seeing channel.c: Nobody there, continuing... in the asterisk full.log. This error is repeated 20+ times per second when it occurs. I thought this problem was specific to one PBX that performs call recording on all the call queues, but after disabling all call recording, the error persists, although less often. The system was hanging badly requiring daily reboots, however since disabling call recording, the system has stabilized. I've since noticed this behavior on another less loaded system. The asterisk versions are 1.2.11 and 1.2.9.1 respectively, and both are running Trixbox. Other systems running older versions of Asterisk, some with AMP/FreePBX don't seem to exhibit this problem. At this point I'm not sure if this is specific to Trixbox, or a problem with later versions of Asterisk. Google turns up very little regarding this error, and the few bugs listed at bugs.digium.com appear to be unrelated. Anyone seen this issue and know what is causing it? Chris ___ --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] Unrecognized frames
Upon investigating a call quality complaint with a conference room, I discovered this error repeated several times in the log. Looking at the source, frametype 5 is An empty, useless frame. Does this indicate an actual problem? app_meetme.c: Got unrecognized frame on channel Local/[EMAIL PROTECTED],2, f-frametype=5,f-subclass=0 All calls in the conference room were via PRI (no voip/sip). The quality complaint was a delay when some parties were speaking, causing multiple people to talk at the same time. I believe this may not have had anything to do with Asterisk (i.e. inbound voip to pstn call), unless frames were being dropped. Chris ___ --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] channel.c: Nobody there, continuing...
I'm seeing this error more and more in the full log. When the error occurs it prints to the log 20+ times per second. At first I thought it was specific to one PBX that performs a fair amount of call recording, but after disabling call recording in all the call queues, the error remains. This system appears to get hung as a result, requiring periodic (daily as of late) reboots. I've since noticed this behavior on another less loaded system. The system versions are 1.2.11 and 1.2.9.1 respectively, and both are running Trixbox. Other systems running older versions of Asterisk, some with AMP/FreePBX aren't exhibiting this problem. At this point I'm not sure if this is specific to Trixbox, or a problem with later versions of Asterisk. Google turns up very little regarding this error, and the few bugs listed at bugs.digium.com appear to be unrelated. Anyone seen this issue and know what is causing it? Chris ___ --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] Issue with g729 codec
Daniel Oakes wrote: Hi All, I have a problem with conferencing, but it's more to do with the g729 codec. I have purchased six licenses for g729 for all our phones, and occasionally want to do conferencing, but at the moment it only allows two people in before the licenses run out. When two people are in the conference and I do a 'show g729' I get the following: *CLI show g729 2/6 encoders/decoders of 6 licensed channels are currently in use And when another person joins the conference they can listen but are unable to speak because all 6 decoders licenses are used up. Any ideas at all from anyone how to fix?? It occurs with all the version I've tried from 1.0.7 to 1.2.1 1.2.7 and 1.2.10. I ran into a similar problem (running out of licenses) today, the result was one way audio. Have you got call recording enabled on any of the extensions that participate in the conference? I have two g729 licenses and ran out of licenses making a single call from a single SIP device. Not sure (yet) why more than two encoder/decoders would be needed to handle a single call with recording. There are similar problems being reported recently on the Trixbox list, but it sounds like they may be related to the Trixbox compile of the latest Asterisk. Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. (831) 621-7928 http://www.scratchspace.com ___ --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] Sound quality issue in one direction and wctdm problem with APIC enabled kernel
I'm chasing down a pop/click type of disturbance on a PBX system. Strangely, the disturbance is only heard by the outside caller, the internal recipient hears the caller crystal clear. This seems to have crept up when upgrading the zaptel driver to the 1.2 series while running 1.0.10. I went ahead and upgraded the entire system to 1.2.4. The system is a ~2Ghz AMD 32bit system, with 512MB of memory and nothing other than Asterisk running. Phone traffic is minimal, perhaps 3 simultaneous calls max, but the problem occurs with just one call. It's located in a data center with ~20ms pings to the ITSP and ~20ms pings to the remote office IP phones. Up to this point, ztdummy was in use without problems, although the timing (zttest) was a hair under the recommended threshold. I dropped in a TDM400P for testing, and although the timing improved, the symptom remained. The system has an IDE drive, and I verified the hdparm dma/irq settings were enabled. The TDM card was sharing interrupts, so I recompiled the kernel with APIC support. Unfortunately the wctdm module will no longer load after recompile and install into the new kernel directory. I went back to the ztdummy driver with the same problem. Below is the relevant errors and info. Chris # modprobe wctdm FATAL: Error inserting wctdm (/lib/modules/2.6.12-prep/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm # dmesg wctdm: disagrees about version of symbol zt_receive wctdm: Unknown symbol zt_receive wctdm: disagrees about version of symbol zt_qevent_lock wctdm: Unknown symbol zt_qevent_lock wctdm: disagrees about version of symbol zt_ec_chunk wctdm: Unknown symbol zt_ec_chunk wctdm: disagrees about version of symbol zt_transmit wctdm: Unknown symbol zt_transmit wctdm: disagrees about version of symbol zt_unregister wctdm: Unknown symbol zt_unregister wctdm: disagrees about version of symbol zt_hooksig wctdm: Unknown symbol zt_hooksig wctdm: disagrees about version of symbol zt_register wctdm: Unknown symbol zt_register # cat /proc/interrupts CPU0 0: 34991774IO-APIC-edge timer 1: 10IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 12:111IO-APIC-edge i8042 14: 170392IO-APIC-edge ide0 15: 383872IO-APIC-edge ide1 18: 0 IO-APIC-level SiS SI7012, SiS SI7013 Modem 19: 164220 IO-APIC-level eth0 20: 0 IO-APIC-level ohci_hcd:usb2 21: 0 IO-APIC-level ohci_hcd:usb3 22: 0 IO-APIC-level ohci_hcd:usb4 23: 0 IO-APIC-level ehci_hcd:usb1 NMI: 0 LOC: 34991738 ERR: 0 MIS: 0 ___ --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] inband dtmf on ploycom ip501?
Damon Estep wrote: Anyone have any hints on how to get the polycom ip501 to send dtmf inband, our upstream providers require inband and the native rfc2833 format of the polycom does not work. In order for inband to make it beyond Asterisk, you need to disable rfc2833 control in the Polycom config file sip.cfg (i.e. via ftp server). tone.dtmf.rfc2833Control=0 It appears that although Asterisk recognizes and uses inband dtmf internally, rfc2833 is used on the external channel. I noticed this behavior when remote IVR systems weren't acknowledging dtmf. Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. (831) 621-7928 http://www.scratchspace.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fxotune fails with valid TDM/FXO card
Mojo with Horan Company, LLC wrote: The recent suggestion on the list was to not use 1.0.9 zaptel You mean the driver, or the version of fxotune? fxotune has been removed from the prior versions of the zaptel driver, it's only included in 1.2 now. As for the driver, is anyone using the 1.2 zaptel driver with Asterisk 1.0.9? The way the downloads are grouped together on the Asterisk web page, I was led to believe they shouldn't be mixed. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cell phone extension woes
I've got a cell phone setup as an extension in a queue. On occasion the cell phone will drop the call due to loss of, or bad, signal. Is there a clean way in the dial plan to reintroduce a call back into the queue when the call is dropped on the extension side? I realize this would occur even during a normal (extension side) call hangup, but as long as asterisk terminates the call when the caller hangs up, this would be fine. Thoughts? Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fxotune fails with valid TDM/FXO card
Fxotune doesn't appear to work with the latest TDM boards. I have a TDM400P rev I card and receive the following when running fxotune : # ./fxotune -i 4 Tuning module 1 Skipping non-TDM / non-FXO Failure! Tuning module 2 Skipping non-TDM / non-FXO Failure! I didn't see anything obvious in the code that ties this to the card revision, but I recalled seeing something on the list about previous changes in this regard. Any suggestions on getting this to work? System details : Fedora Core 4 Kernel 2.6.13-1.1526_FC4smp Asterisk CVS-v1-0-10/02/05-15:54:21, Copyright (C) 1999-2004 Digium. Zaptel 1.0.92 dmesg : Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 0 (United States / North America) Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. (831) 621-7928 http://www.scratchspace.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Resolving QOS problems
I'm looking for advise on troubleshooting QOS problems. After much searching and reading online (Google, Voip-Info Wiki, etc.) I don't feel any closer to finding the right tools to solve my problem. Any info you would like to share would be much appreciated, and I'm sure the thread will server others in the future. The problem : - I'm having intermittent problems with the audio cutting out on calls. At the same time the audio problems occur, I often see these in the full log : Received iseqno 122 not within window 123-123 These range from sounding like bad cell phone calls, to the audio track cutting out in one or both directions for up to 20-30 seconds. I also see dropped calls that seem to be a result of the IAX connection going away. The environment : - I've got an * server located at a data center with good connectivity, 10 hops to my IAX provider, and ~34ms ping times. They (IAX provider) use Cogent which concerns me a bit, but I'm not ready to jump to conclusions just yet. My IP phone is connected via enhanced DSL (static addresses, no PPPoE) and I'm 12 hops away from my * server. My DSL provider has direct connectivity and peering agreements with the data center my server is located in. I've set QOS priority on the LAN port (Linksys router) the phone is connected to, and I've dropped the MTU to 576 as suggested for lower speed links. (1.5Mbs/384kbps in my case). Both these changes seemed to make an improvement over previous calls. Currently I don't believe the bulk of my problems to be between the phone and the * server. testyourvoip.com tests consistently show a 4.4 score (the maximum for ulaw) and rarely shows errors. Ulaw is the codec used for both the SIP calls and IAX trunk. What I'm looking for : -- I'm trying to determine the cause and location of the problem between my * server and the IAX provider (and possibly my IP phone), and see what if anything I can do to reduce the occurrence of these drop outs. I'm looking for a couple of things : 1. A method of monitoring RTP/IAX traffic QOS at the PBX in real time. 2. Tools that might be used to determine the location of the problem. I.E. An RTP/IAX traceroute tool. What I'm hoping to find is something that either integrates directly with *, or captures live RTP/IAX traffic and provides real time statistics on calls. What I've found : - I saw Telchemy's VQMON_EP product, but it's unclear how it would work with Asterisk. Many other companies in this market seem to leverage off of Telchemy's products. http://www.telchemy.com/partners.html http://www.voiptroubleshooter.com/tools/voiptr_tools.htm All of the products above seem to be aimed at large enterprises with deep IT pockets. I wouldn't mind ponying up a reasonable sum for a tool that does the job, but I lack the time to thoroughly evaluate everything that may be out there. I haven't found much on the open source front. I've seen Windows RTP Quality Monitor which might be useful, but it's beta and hasn't been updated in over a year. It seems to me that Ethereal might be integrated with a graphical tool, and if nothing else provide postmortem statistics on a phone call. Request for comments : -- What are people using to troubleshoot these problems? What commercial software works for you? What open source projects are you using? Do you have suggestions on projects that might be glued together to provide this functionality? Thanks in advance. Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stopping retransmission on messages
I'm seeing a number of these logged in full while my * system is idle, but I haven't found a good description of what they mean. Can someone oblige? I have a single SIP phone registered and an IAX trunk. Chris Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 732: Found Sep 19 22:13:44 DEBUG[18720]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 732: Found Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 733: Found Sep 19 22:13:44 DEBUG[18720]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 733: Found Sep 19 22:13:44 DEBUG[18720]: Registration successful Sep 19 22:13:44 DEBUG[18720]: Cancelling timeout 1882 Sep 19 22:15:29 DEBUG[18720]: Scheduled a registration timeout # 1886 Sep 19 22:15:29 DEBUG[18720]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 734: Found Sep 19 22:15:29 DEBUG[18720]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 734: Found Sep 19 22:15:29 DEBUG[18720]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 735: Found Sep 19 22:15:29 DEBUG[18720]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 735: Found ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP providers -- California, U.S.
jennyw wrote: Hi, Just wondering if people could suggest a good VoIP provider that can service the San Francisco Bay Area and the Los Angeles area. I've tried race.com (recommended to me) but they're kind of hard to get ahold of. Any other suggestions? This is for a business, so reliability is key. I did see the recent thread about this, and while I saw a few mentioned, I didn't see anything about how reliable the different vendors are, or whether people are using them for business or personal use. I've had good luck so far with Teliax. Difficult to get on the phone some times, but support response has been fast via email and the system has been rock solid in my testing so far. I've seen other folks give them good marks as well. Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. http://www.scratchspace.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP trunk rollover problem
Hello,
I've got an Asterisk system with 3 SIP trunks configured. Each SIP
trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound
call routing (via AMP 1.10.007a) uses the 3 trunks in descending order,
all set with max channels to 4. Unfortunately, when the first trunk
reports a 480 Service Unavailable (all ports in use), Asterisk reports
congestion without rolling to the next available trunk.
I've looked at the AMP dialplan on this system, as well as a more
recent system (1.10.009beta1), and although the dialplan has been
improved, it still doesn't seem to account for this condition. As you
can see below at step 7, if the max channels have been used on the
current trunk, the call fails.
What is the correct way to do fail over between trunks, and in an AMP
friendly way that won't get clobbered during the next config change?
Regards,
Chris
[macro-dialout-trunk]
exten = s,1,GotoIf($[foo${ARG3} = foo]?3:2)) ; arg3 is pattern password
exten = s,2,Authenticate(${ARG3})
exten = s,3,Macro(user-callerid)
exten = s,4,Macro(record-enable,${CALLERIDNUM},OUT)
exten = s,5,Macro(outbound-callerid,${ARG1})
exten = s,6,SetGroup(OUT_${ARG1})
exten = s,7,CheckGroup(${OUTMAXCHANS_${ARG1}})
; if we've used up the max channels, continue at 110 (n+101)
exten = s,8,SetVar(DIAL_NUMBER=${ARG2})
exten = s,9,SetVar(DIAL_TRUNK=${ARG1})
exten = s,10,AGI(fixlocalprefix) ; this sets DIAL_NUMBER to the proper
dial string for this trunk
exten = s,11,SetVar(OUTNUM=${OUTPREFIX_${ARG1}}${DIAL_NUMBER}) ;
OUTNUM is the final dial number
exten = s,12,Cut(custom=OUT_${ARG1},:,1) ; Custom trunks are prefixed
with AMP:
exten = s,13,GotoIf($[${custom} = AMP]?16)
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM}) ; Regular Trunk Dial
exten = s,15,Goto(s-${DIALSTATUS},1)
exten = s,110,Noop(max channels used up)
exten = s-BUSY,1,NoOp(Trunk is reporting BUSY)
exten = s-BUSY,2,Busy()
exten = s-BUSY,3,Wait(60)
exten = s-BUSY,4,NoOp()
exten = _s-.,1,NoOp(Dial failed due to ${DIALSTATUS})
___
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[Asterisk-Users] 64 Bit Support?
I'm running * on an AMD 64 system with FC3 x86_64, everything works fine so far. Programs can be rewritten to take advantage of the the 64 bit architecture and the extra computing power. Having seen that many high end systems are using 32 bit Xeon based systems for call capacity, I'm wondering if/when we might see a * port for 64 bit, and if there will be a substantial difference in call capacity. AMD's 64 bit systems are comparable in price, in some cases cheaper, and in benchmarks are often more powerful than the Intel counterpart (depending on what is running). Thoughts? Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] /usr/bin/ld error on make asterisk with Fedora Core 3
Dave Green wrote: I've downloaded the latest CVS as of yesterday. Zaptel and libpri compile and link OK but after issuing the make asterisk command I get the following: /usr/bin/ld: cannot find -lidn collect2: ld returned 1 exit status make[1]: *** [app_curl.so] Error 1 make[1]: leaving directory '/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Linux version is 2.6.9-1.667 gcc version 3.4.2 I compiled an earlier version of asterisk OK on a RH9 install but couldn't get zaptel to compile and I didn't have the kernel source files for RH9, so went to FC3. One step forward, one step back :) Any pointers to fixing / working around this error ? Works fine for me, did you su - to pick up root's env? It's saying it can't find the idn library. Make sure you have libidn-0.5.6-1 installed. Also, you need to update that system, there's a kernel update for a nasty (albeit local) security hole. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] make clean DO IT!
Christopher L. Wade wrote: Andrei (MPI) wrote: Brian West wrote: Just an FYI to all out there that are upgrading after this weekend's run of CVS updates that are in now... MAKE SURE YOU DO make clean. If you don't and asterisk acts funny this is why. Anytime any struct like ast_channel (which was changed over the weekend) and you don't make clean you'll end up with an asterisk box that acts retarded. So please before reporting a bug do a fresh checkout or make clean and try again. Also, do not forget to: rm -rf /usr/lib/asterisk/modules -- Andrei (MPI) great reason for a 'make uninstall' or the like? removes everything related to * except config files? - dont mean to start a different thread, it would just be a nice feature - like the 'make update' that just got added to zaptel. Isn't that what package managers are for? If we weren't using CVS to stay on the bleeding edge, it would be nice to have a reasonably current Asterisk RPM. Hey, looks like there's a 1.0.3 FC3 RPM available here : ftp://ftp.linuxsys.com/pub/LSE/packages/Fedora/SRPMS/ My $.02 ;-0 Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Out the box solutions?
Lane wrote: Hi, again. I've spent a week trying to get asterisk to work on FreeBSD unix, with some success. Everything works until I plug the box into the TELCO line and then the line goes off-hook and stays that way. I wasn't able to get the zaptel stuff working under 5.3, but that has more to do with me running the amd64 port, but I do understand it works. I think the FreeBSD port is a little behind the Linux stuff. So I bit the bullet and decided to install the application on a fresh linux install. Not to start an OS war, here, but linux is ... difficult ... for an old unix hand to get his mind around. It's a completely different landscape! I've been a BSD guy for years and I had not really been interested in learning Linux, but everyone seems to want to see that on your resume. I finally got the opportunity to start working with RedHat 7.3 in production about a year ago (not for * mind you). I also used Red Hat Enterprise and played a little with Fedora. All I can tell you is it will be easier if you _embrace_ Linux and learn it. I find things much easier to resolve now that I understand how things are wired under the hood. I'm still a BSD guy, but as someone else pointed out there are pros and cons to both. But I digress. I chose FC3 (Fedora Core) for the install, and now I'm sorry that I did. Well I must say that the Core 3 release is much closer to production stable than any of the previous releases, I'm pretty impressed with how well the OS is put together in comparison to RHES (which sort of comes from Fedora anyway). I've gotten everything to work so far, but it did take some reading and really had more to do with learning * than Fedora. At least with unix I was able to get a dial tone! Not so much with this flavor of linux. Each time I run modprobe wcfxs I get the following errors in /var/log/messages: The problem seems that you may not have updated your system, this is a udev issue that's fixed in an update. If you're not familiar with up2date and this is a test system, you might want to install a full system and use Gnome (like windows) or KDE so you can get exposure to the system. You also need to modify the init.d script to load the modules with the correct *mod* commands (like you said, all Linuxes are a little different). Here's a good pointer page for FC3, I wish I saw it before I figured it out myself... http://voip-info.org/tiki-index.php?page=Asterisk%20Fedora%20Core%203#comments I'm not so interested in notifying these guys at lists.sourceforge.net, since I'm only interested in running asterisk. They already know and fixed it : http://lwn.net/Articles/111858/?format=printable So ... the question: What flavor of linux does asterisk actually run on Out the box? None. As a BSD guy you should know nothing just runs out of the box ;-) Although ports does a good job. I'm not scared to compile asterisk, but I'm not at all interested in recompiling a linux kernel. Not needed, you just need to download the SRPM (that matches the updated kernel, at this time is kernel-2.6.9-1.724_FC3) install it (rpm -ivh kernel-2.6.9-1.724_FC3.src.rpm) and follow the instructions on the wiki page above. All you're doing is the equivalent of make depend to update/create a header file. Of course if that is the only way, then I guess I'll just bite another bullet. Keep an open mind, that's all I can say. At this point I like the RedHat based Linux distro. Others are also good, some more suited to desktop or server, but I believe that RedHat is one of the leading distros in terms of community/commercial support, much the same way FreeBSD is over Net/Open BSD. Hell! I want this PBX to work so bad that I can almost taste it! Been there, I've got dialtone now and have both TDM400P FXO/FXS ports working. Chris ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems with udev on FC3
I had the same problem and I see that this was addressed in udev-043 : http://lwn.net/Articles/111858/?format=printable (search for zaptel) FC3 has udev-039-8.FC3 installed by default. If you run up2date, an update to udev-039-10.FC3.6 is available that fixes this problem. Also the typical zaptel entries don't show up in /var/log/messages during boot when things are installed correctly. IMPORTANT: You may need to force install this rpm (at least in the graphical up2date client) if you already modified : /etc/udev/rules.d/50-udev.rules If you don't update the new copy of this file, your system won't boot and you'll have to boot single user to fix it. On a side note I also found this page which discusses udev and how to workaround device issues. http://fedora.redhat.com/docs/udev/ Chris Original Message below --- I've been testing * on FC3. I have everything compiled and installed. However, when I do 'modprobe wcfxo' (I have an X100P clone), I get the following in /var/log/messages: Nov 24 10:23:40 jfd wait_for_sysfs[3366]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zaptimer' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to linux-hotplug- devel at lists.sourceforge.net Nov 24 10:23:40 jfd wait_for_sysfs[3368]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zapchannel' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to linux-hotplug- devel at lists.sourceforge.net Nov 24 10:23:40 jfd wait_for_sysfs[3370]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zappseudo' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to linux-hotplug- devel at lists.sourceforge.net Nov 24 10:23:40 jfd wait_for_sysfs[3372]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zapctl' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to linux-hotplug- devel at lists.sourceforge.net Nov 24 10:23:44 jfd wait_for_sysfs[3377]: either wait_for_sysfs (udev 039) needs an update to handle the device '/class/zaptel/zap1' properly (no device symlink) or the sysfs-support of your device's driver needs to be fixed, please report to linux-hotplug- devel at lists.sourceforge.net I added the udev rules as described in README.udev. Any ideas? TIA -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/4da29c87/attachment.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Bruno Hertz wrote: On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote: I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. The problems you had and were resolved with ztdummy, were they primarily IAX related ? Since, after all, the main channels relying on special timers are Meetme, IAX and (maybe) MusicOnHold according to http://www.voip-info.org/wiki-Asterisk+timer Just want to be sure, since I still believe my mere demo playback issue likely has a different reason ... I'd like to chime in here as I have a similar problem. I have been toying with * on other (cheapo) hardware not so successfully (mainly due to the audio chipsets). I just purchased an ASUS AV8 (Socket 939 Athlon 64 3500+) system for my real world testing, it's a high end MB and overall it has 98% of the feature set for what I wanted to accomplish. Currently I'm running FreeBSD 5.3 under the amd64 port of the OS (fyi). I'm experiencing the exact same symptoms - choppy clicking of the demo voice. I'll start by saying that I have done a reasonable amount of research on *, MB chipsets, and FreeBSD, and I've spent considerable time getting the basic functionality to work. The ports version of * under FreeBSD needed some tweaking to work under amd64 vs i386, but I have a working version including h323 and oss that works with the demo stuff. From what I have read the issue with choppy sound under the demo voice seems to be due to a timing issue, one that can't be solved under FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as far as USB stuff that will handle this. I do not have a Digium card installed yet, but I will have a TDM400P in a couple of days. Will a Digium card with the current driver solve the problem ? (zaptel doesn't compile for FreeBSD 5.3 amd64, maybe for i386). Given that I have a working installation with the same symptoms as reported, I'm leaning towards us having the same problem. If this is a timing issue, it would be great to solve this in a systematic way (without external hardware). Thoughts? Chris ___ 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
