Re: [asterisk-users] Strange Issue: asterisk deleted
- Original Message - Hi Thank you for your support. The server is actually compromised, I discovered that after making a deep trace using the audit daemon and looking for the kill signal (SIGKILL) that terminates asterisk. I discovered that there is an executable with a random name in the /boot folder that is killing and deleting asterisk !!! This executable is launched by a service in /etc/rc.d/ with the same random name. When I stopped this service, a new service was created with another different random name and it too is killing and deleting asterisk. This was the evidence i needed to be convinced that the server has a virus and is compromised. The good thing is that this is a fresh install and hence there are no sensitive data or a lot of work done on it so i will reinstall the OS and start over. The bad thing is that I spent more than 4 days trying to understand what was going on. Very interesting. Any ideas on how the system was compromised? Are any other daemons being actively replaced, or just Asterisk? I did hear of a similar issue to the one you describe (also on an Asterisk box) via a third party recently, but don't have any real specifics other than it being Asterisk 1.4.x on Debian (5 or 6), running on a local LAN, no outside access. Curious if there are any commonalities to the two compromised systems. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 10/22/2014 03:55 PM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim I can't help with your root problem (maybe check core show function FAXOPT?), but the spandsp site is up. Try using www.spandsp.org. Downloads are available here: http://www.spandsp.org/downloads/spandsp/ It is up now, thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 23/10/2014 3:55 AM, Tim Nelson wrote: - Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) No thoughts on your problem, I do think you will need a newer version of spandsp through - the site seems to be up now. The version of SpanDSP is not in question at this point. The problem lies in I need a way to use the T38 Gateway function, but *also* initiate the reinvite to T.38 on the call as the provider will not do this, saying it is the *caller*'s responsibility. This is contrary to past experience however... --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 23/10/2014 10:07 PM, Larry Moore wrote: On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? Larry. Have you had a look at https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance As an exercise you could disable T.38 on 'Asterisk calling system', if you have an ATA which is originating the call to 'Asterisk calling system' disable T.38 on that device too and disable in your sip.conf using t38pt_udptl=no. If you are using SendFax() on 'Asterisk calling system' ensure T.38 is not able to be used. If using an ATA connecting to 'Asterisk calling system' ensure you have set in your peer's configuration canreinvite=no or directmedia=no, depending on the version of Asterisk you are running on this system. On Asterisk system in '(box in question)' set directmedia=no for the peer which is connecting to 'SIP Provider' and also to 'Asterisk calling system', you may want to set setvar=FAXOPT(gateway)=yes in your peer config to 'SIP Provider' otherwise it will need to be set in your dialplan. Set your verbose debug to at least 3 on '(box in question)', possibly a little higher and send a fax - you may now see the Fax Gateway detect CED. Not sure if this is suppressed in You may want enable udptl debugging on '(box in question)'. I do *not* want to disable reinvites or udptl media as it is required for T.38 operation. All testing shows (via packet capture) no reinvite for T.38 is happening on the call leg with the ITSP. Thank you for the idea however on setting the FAXOPT for gateway in the provider SIP peer definition, I will test that shortly. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - On 22/10/2014 11:23 AM, Tim Nelson wrote: Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: What type of endpoint are you using which is originating the call and is it T.38 capable? The originating endpoint is an IAXmodem controlled by Hylafax. Actual call flow is IAXmodem --G.711u via localhost-- Asterisk (old version with no T.38 support) --G.711u-- Asterisk 11.x --G.711u/T.38-- ITSP The problem lies on the Asterisk 11.x system not being able to reinvite to T.38 on the call leg with the ITSP, and given the ITSP does not do this either, the call is stuck in G.711u with varying performance. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
- Original Message - Greetings- Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2]. Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html *bump* Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as gateway do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. :) Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_fax T.38 Gateway with SpanDSP - Force ReINVITE?
Greetings- Working with the T.38 gateway functionality that is sparsely documented [1] , I'm attempting to get the following functional: Asterisk calling system - Asterisk system in T.38 Gateway Mode (box in question) - SIP Provider The problem is: -The provider is not initiating a reinvite to T.38, even though it is 100% supported -Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected) So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2] . Thank you, --Tim [1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway [2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html -- _ -- 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] Polycom DND + Intercom/Paging Override?
- Original Message - Tim, I THINK but I'm not sure that you can do this with the Polycom multicast page function. Have you attempted this yet? Thanks david Given the odd nature of multicast paging with Polycom, I was hoping to avoid such a setup. My recollection is having this work previously with an older version of Asterisk (1.4.x?), and the same handsets. Time to check archived backups... Thank you for the suggestion though, I may have to go that route. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom DND + Intercom/Paging Override?
Greetings- As many of your are Polycom experienced, I was hoping some kind soul could provide direction on a specific issue. On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding an instance where, using intercom/paging functionality of FreePBX, I need to override an end user's 'Do Not Disturb' selection on the handset. By default, DND simply rejects all inbound SIP INVITEs. However, a page/intercom needs to be allowed through. Any suggestions? I've read reports this is doable using Polycom config options for call priorities, but I've had no such luck yet. Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
- Original Message - On 13 Feb 2014, at 18:10, Tim Nelson tnel...@rockbochs.com wrote: I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Just because Box B was receiving 65MBps doesn’t mean box A was sending them. I suspect it’s probably the same one repeated, due to some kind of network problem. Do you have a pcap so you can look for the ID in the packets to see if they are the same? Would be good if you can prove A sent them too (traffic stats from SNMP monitoring or something). Right, but a packet capture shows the source to be box A, and the destination to be box B. NMS reports from the same time period confirm the traffic flows. I'm not guessing or stabbing in the dark, I did my homework before posting. :) Checking the IDs across ~25 packets, all have different SIP IDs. Any thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP OPTIONS storm?
- Original Message - SIP options message is due to check the peer registration is keepalive. As per my understanding it might be because of network flap may be wireshark trace can give you any clue. Regards Correct. I understand the role and function of the OPTIONS requests. The issue is why was Asterisk sending out 65Mbps worth of them to one peer? I did get a capture of the traffic, but nothing appears to explain *why* the traffic was there to begin with. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP OPTIONS storm?
Greetings- I recently experienced an odd situation. I have an Asterisk 11.5.0 system (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is not set (aka default of 60secs). Of course, logs on Box A were not set to show debug info, so there is no indication of a problem. Logs on Box B show no issues, only at a very specific start time, there are suddenly tons of: [2014-02-13 00:12:50] DEBUG[31516] chan_sip.c: Allocating new SIP dialog for 2a338cf5518531e31190bd4b7826d137@x.y.z.166:5060 - OPTIONS (No RTP) I've done quite a bit of searching, but am not finding anything of consequence. Also, the Asterisk changelogs are not providing anything that would indicate this is known and fixed, at least for the 11.x branch. Thoughts/suggestions? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CTI
- Original Message - http://camrivox.com/products/flexor-cti-salesforce/ We've used this for a few clients. How were your experiences with it? I have a customer that will want this type of integration in the near future, and would love to hear how installation, operation, and support has been. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multi-Voicemail Message?
Greetings- I have an odd scenario where I need to dial an extension (lets call it 555), the system prompts for a list of voicemail boxes, then once complete, allows the caller to leave a voicemail that is sent to all voicemail boxes previously specified. How would you do this? Obviously calling Voicemail(), but how to get input for multiple extensions/voicemails, and delimit them properly for passing to Voicemail()? All ideas welcome. Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstDB Partial Replication?
- Original Message - Is anyone aware of a way to replicate parts of the AstDB to another Asterisk install? For example, to export all CF entries on a FreePBX based system to another system running FreePBX, I might do: asterisk -rx 'database show' | grep CF This gives me a list of data, which I can rsync to another host to reimport using 'database put'. BUT, the problem comes in when I want to sync CF entries to/from both Asterisk systems. I seem to be having race conditions where an entry is removed on system A, but before that removal can sync to system B, we've already imported that to system A again. Does this make sense? TLDR; How do I sync AstDB entries between two hosts, in both directions, while maintaining data integrity? Any takers? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AstDB Partial Replication?
Is anyone aware of a way to replicate parts of the AstDB to another Asterisk install? For example, to export all CF entries on a FreePBX based system to another system running FreePBX, I might do: asterisk -rx 'database show' | grep CF This gives me a list of data, which I can rsync to another host to reimport using 'database put'. BUT, the problem comes in when I want to sync CF entries to/from both Asterisk systems. I seem to be having race conditions where an entry is removed on system A, but before that removal can sync to system B, we've already imported that to system A again. Does this make sense? TLDR; How do I sync AstDB entries between two hosts, in both directions, while maintaining data integrity? Thanks --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange issues with newly rebooted machine
- Original Message - No, my phones aren't getting a response from the server. I can't even get any output from the server if I do: sip show peer name load This command usually loads the peer from the db and shows me it's configuration. In this case, I get nothing. I do have rtcachefriends=yes in my sip.conf. In fact, this server has a virtually identical configuration to one that is already running. (I sync the configurations using unison.) I don't THINK this is a configuration issue. Any ideas, though? It sounds like Asterisk is hung in general. Next step, stop asterisk altogether, edit your /etc/asterisk/logger.conf to output all to a logfile: full = notice,warning,error,debug,verbose,dtmf Then, do a 'tail -F /var/log/asterisk/full', and startup Asterisk. I'm guessing you'll be able to see some errors flow by, but more importantly, maybe the log will stop, showing you exactly what is hanging. Good luck! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI - Tickless Kernel?
Greetings- I'm running some USB DAHDI hardware on a system with a tickless kernel. The audio quality is quite poor. Could the tickless kernel be to blame? If so, when recompiling a kernel that is *not* tickless, is there a recommended KERNEL_HZ value? IIRC, older kernels used to be 1000, but newer ones are 250. Thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Soundcard - Recording?
- Original Message - On Tuesday 28 May 2013, Tim Nelson wrote: Greetings- I've got a curious project that I could use some input on. I'd like to use Asterisk to record some audio channels via USB 'soundcard'. When audio passes through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and write it to a wav file. I'm perfectly competent with the dialplan portion of the recording, but I don't know about the following: -How does Asterisk know a new audio stream/source is beginning/ending? -Can I have more than one CONSOLE device -Is ALSA or OSS the preferred audio system with Asterisk these days (Asterisk 11 presumably) Any thoughts? Or, do you have any alternative ideas that would work better than using Asterisk for this? Do you need to use Asterisk for this? What's wrong with using the rec command? $ rec -b8 -r8k -c1 file01.wav That is a good suggestion, but of course I'd need to add some 'magic' for control of the recording at various intervals, after silences, etc. Also, please -- and don't take this too personally, because you are by no means the only guilty party; this applies to other users as well, and they know who they are -- when asking questions on this list, please try not to come across sounding furtive or evasive. Not only is it easier to answer a question with too much information than too little, but a question asked with such an air of secrecy about it actually dissuades people from offering help. It was not my intent to come across this way. To be honest, the project I'm working on isn't even completely fleshed out, and I too am slightly 'in the dark' on details. I was simply doing some preemptive investigative work into the software side of things. It looks as though you are deliberately concealing information because you don't trust us not to steal an idea for which you are about to take the full credit. This is contrary to the very spirit of the mailing list, which is all about sharing information, and people will naturally close ranks to squeeze out that sort of behaviour. Your position is understood, but incorrect in this case. And, again, there is no deliberate concealment, only a lack of details on my part. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Soundcard - Recording?
- Original Message - You are still being a bit evasive but should I understand that you want to run a headless machine with open microphones that records what ever it hears? What do you want to do with each sound bite? How long does the silence have to be before you close the recording and dispose of it (save, e-mail, upload, whatever). Sounds like a security monitoring package (minus the video) should do the job? Thanks for the info, I will have a look at those. A little Googleing shows up these. http://oreka.sourceforge.net/about/ http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html What else do you want it to do? No idea at this point. I'm just doing some preemptive legwork for a project coming down the pipe. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Soundcard - Recording?
- Original Message - I'll take a stab, since you said no GUI and also USB based mic. Raspberry Pi project? I'm interested in this vein as well, Nope, the RPi is not a component. BUT, a good suggestion for a small board. I'll keep it in mind. especially after the recent post about voice recognition. I was thinking that Raspberry Pi's with mics could live around my house and all have dedicated always-open channels to a conference bridge in the main asterisk box. I was planning on using ALSA and a USB mic on a local Raspberry Pi asterisk instance. Very interesting. You could use the CONSOLE for the audio device and at the same time get the ability to make/take calls via Asterisk CLI or .call files. So given that we know basically what you are trying to do, the original question was OSS versus ALSA for USB mic, correct? Has anyone had any thoughts on that? I thought ALSA was built in to the kernel and OSS required some hacks. But that is a pretty fuzzy recollection. I think it depends on the kernel being used. Research is needed, but IIRC, you are correct, ALSA is included or at least fully supported whereas other sound systems are 'less than stellar'. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Soundcard - Recording?
Greetings- I've got a curious project that I could use some input on. I'd like to use Asterisk to record some audio channels via USB 'soundcard'. When audio passes through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and write it to a wav file. I'm perfectly competent with the dialplan portion of the recording, but I don't know about the following: -How does Asterisk know a new audio stream/source is beginning/ending? -Can I have more than one CONSOLE device -Is ALSA or OSS the preferred audio system with Asterisk these days (Asterisk 11 presumably) Any thoughts? Or, do you have any alternative ideas that would work better than using Asterisk for this? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Soundcard - Recording?
- Original Message - What are you trying to accomplish? What is the USB 'sound card' attached to? Your description is too cryptic for someone to propose a solution. The target use is to record mic level audio from various devices (could be an omnidirectional room mike, phone handset, etc). --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Soundcard - Recording?
- Original Message - Sorry for the blank message. Fingers pressed send while brain was disenaged. Would Audacity be a better choice? http://wiki.audacityteam.org/wiki/Multichannel_Recording It would absolutely be a better solution. However, the recording is to be automated on a small system with no GUI, only console/SSH access. As such, running a full featured audio recording/mixing application in realtime (with user control) is not an option. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2
First thing to *ALWAYS* check is if you have any Asterisk version specific modules (Fax for Asterisk, G.729, etc). Ensure these are not loaded (noload in modules.conf, or simply move them out of the asterisk modules dir). Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 - Original Message - After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root@localhost asterisk-11.1.2]# asterisk -vvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/logger.conf': Found == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree == Registered custom function 'MESSAGE' == Registered custom function 'MESSAGE_DATA' == Registered application 'MessageSend' == Manager registered action MessageSend == Manager registered action DataGet == Parsing '/etc/asterisk/codecs.conf': Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found == Parsing '/etc/asterisk/dnsmgr.conf': Found [2013-01-10 14:20:10] ERROR[27062]: config_options.c:512 aco_process_config: Unable to load config file 'acl.conf' == Parsing '/etc/asterisk/http.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Login == Manager registered action Challenge == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action GetConfig == Manager registered action GetConfigJSON == Manager registered action UpdateConfig == Manager registered action CreateConfig == Manager registered action ListCategories == Manager registered action Redirect == Manager registered action Atxfer == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action PresenceState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Manager registered action SendText == Manager registered action UserEvent == Manager registered action WaitEvent == Manager registered action CoreSettings == Manager registered action CoreStatus == Manager registered action Reload == Manager registered action CoreShowChannels == Manager registered action ModuleLoad == Manager registered action ModuleCheck == Manager registered action AOCMessage == Manager registered action Filter == Registered custom function 'AMI_CLIENT' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_humbug.conf': Found [2013-01-10 14:20:10] NOTICE[27062]: manager.c:7545 __init_manager: Invalid keyword displaysystemname = yes in manager.conf [general] == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/cdr.conf': Found [2013-01-10 14:20:10] NOTICE[27062]: cdr.c:1613 do_reload: CDR logging disabled, data will be lost. -- CEL logging disabled. == Parsing '/etc/asterisk/udptl.conf': Found [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1413 removed_options_handler: t38faxudpec in udptl.conf is no longer supported; use the t38pt_udptl configuration option in sip.conf instead. [2013-01-10 14:20:10] WARNING[27062]: udptl.c:1415 removed_options_handler: t38faxmaxdatagram in udptl.conf is no longer supported; value is now supplied by T.38 applications. Asterisk PBX Core Initializing Registering builtin applications: == Registered custom function 'EXCEPTION' == Registered custom function 'TESTTIME' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [ExecIfTime] == Registered application 'ExecIfTime' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ImportVar] == Registered application 'ImportVar' [Hangup] == Registered application 'Hangup' [Incomplete] == Registered application 'Incomplete' [NoOp] == Registered application 'NoOp' [Proceeding] == Registered application 'Proceeding' [Progress] == Registered application 'Progress
[asterisk-users] Asterisk 1.8.19.0 - [2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL
I'm getting this error message on my Asterisk CLI, and in the logs, roughly every 10-20 seconds: [2012-12-18 19:19:51] ERROR[24485]: astobj2.c:115 INTERNAL_OBJ: user_data is NULL While it doesn't appear to be actually affecting anything, I'm curious to know what the error represents, where it's coming from, and of course, if there is a fix for it. All info appreciated, thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] Polycom IP450 Firmware Issues
- Original Message - Tim, What version are you on? There is a specific upgrade path for pre 3.3. Yes, that was the issue. I needed to upgrade to version 3.3 first, then upgrade to latest 4.x was no problem. Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [OT] Polycom IP450 Firmware Issues
I have a site with Polycom handsets on all the desks, mostly IP650s, some IP550s, and some IP450s as well. I need to update the firmware on the IP450s. However, the firmware simply won't load. The latest firmware (4.0.3 Rev F) supports all phones at this site, and was downloaded from here: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html The phone pulls the firmware from the PBX via TFTP (as expected), but always results in 'Error: Image is not compatible with the phone'. As a troubleshooting step, *ALL* firmware has been removed from the TFTP root, and *ONLY* the new firmware placed there. So, is the Polycom firmware matrix wrong about this phone/firmware compatibility, or am I missing something? The bootrom has also been upgraded to the latest without any problems. Thoughts? My head is getting sore from banging it on my desk... :/ Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 -- _ -- 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] watchdog like functions
- Original Message - Switching to SIP is likely your best solution. IAX is buggy. Always has been, and I'll bet always will be. Alright, I'll bite on this one. Can you give any specifics about IAX being buggy, other than throwing out random claims? I understand it doesn't get the industry use and acceptance SIP has seen, but that doesn't automatically discount it's functionality correct? If anything, I've found SIP to be more finicky, mostly due to far end NAT issues or general interop problems. I guess I'm just curious about your IAX experience that would lead you to discount it as 'buggy'. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] watchdog like functions
- Original Message - I wish to ask if there is way to keep IAX trunk connection up. I have a small server on Xen VPS but notice that my IAX trunk drops after some time. I understand there is cron job to function as sip watchdog. My asterisk is 11.0.1 You'll want to use 'qualify=yes' for your IAX2 peers which keeps registrations active by sending a 'ping' every 60 seconds (by default). Quite a bit of detail available here: http://www.voip-info.org/wiki/view/Asterisk+iax+qualify --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP - Authenticated vs Unauthenticated Calls
- Original Message - Tim Nelson wrote: Greetings- Hola, I'm running into an issue as follows, in simplified form: A remote Asterisk box, when registered/peered via SIP to a central server, and makes a call to that central server, is *sometimes* authenticated and calls go through properly (via from-internal context), and *sometimes* is unauthenticated, and all calls are greeted with congestion() via the from-sip-external context. Yes, as you can tell, FreePBX is in play here too. Grabbing captures of a working call vs a non-working call, I'm seeing on the working call, the central Asterisk server is responding to the INVITE with a 407 Proxy Authentication Needed, box responds, call goes through. On the non-working calls, the central Asterisk server is responding with a simple 100 Trying, then 200 OKs the session as it throws it into from-sip-external assuming the box is not authenticated. So... and pardon my rambling above... why is this the case? In what circumstances would Asterisk respond to the same peer differently, seemingly at random? chan_sip can match to a peer a few different ways: 1. The user portion of the From header matches a configured peer in sip.conf 2. The received IP address/port matches a configured peer in sip.conf using insecure=very or combination thereof. It's possible that you are relying on #1 but not explicitly overriding the user portion in the calling Asterisk using fromuser. Without doing this the user portion carries caller ID number information, with can obviously change between calls. That's my best guess without sip set debug on output for a non-working call and the configuration. Thanks Joshua- In this case, we're using SIP registration to peer the remote systems to the 'central system'. In option #1 above, the 'user' portion is always the CID we set for the outbound call, but the actual SIP user is something different like 'site12' for example. So, it would appear #1 is not a match... That leaves us with option #2. We're using 'qualify=yes' on both sides of the SIP peering. If a peer becomes unreachable (fast UDP state table timeout on a remote firewall for example) as seen by the central system, and an outbound call is made from the remote system, that would mean the call is coming from an unknown IP:port. Would this then make sense Asterisk would simply throw it into the from-sip-external context as an unknown/unauthenticated call? And of course, when the peer *is* registered, and a call is made, Asterisk on the central system allows the call as authenticated due to the source IP/port being known via the registration status? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP - Authenticated vs Unauthenticated Calls
Greetings- I'm running into an issue as follows, in simplified form: A remote Asterisk box, when registered/peered via SIP to a central server, and makes a call to that central server, is *sometimes* authenticated and calls go through properly (via from-internal context), and *sometimes* is unauthenticated, and all calls are greeted with congestion() via the from-sip-external context. Yes, as you can tell, FreePBX is in play here too. Grabbing captures of a working call vs a non-working call, I'm seeing on the working call, the central Asterisk server is responding to the INVITE with a 407 Proxy Authentication Needed, box responds, call goes through. On the non-working calls, the central Asterisk server is responding with a simple 100 Trying, then 200 OKs the session as it throws it into from-sip-external assuming the box is not authenticated. So... and pardon my rambling above... why is this the case? In what circumstances would Asterisk respond to the same peer differently, seemingly at random? I'm happy to provide any details required, but I'm having a brain freeze on what would be relevant at this point. Thanks for any pointers or ideas! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Odd Sangoma Card Issues
- Original Message - Has anyone seen issues with recent Sangoma T-1 cards and Sangoma Analog cards on multiple different servers? On T-1: we get NO traffic, no interrupts, and no increase in number of packets and the PRI does not come up. On Analog: The ports do NOT go red when you unplug the phone line from FXO ports, can't dial out of the ports either. We have installed 50+ analog cards and almost as many T-1 cards and we never had these issues before. We have a ticket in with Sangoma but they want to blame the line or Asterisk. We utilize Sangoma interface cards almost exclusively and have had no problems whatsoever including several new cards this week put into production. I have to assume if there were some sort of issue inherent to their hardware the various channels (mailing lists, forums, etc) would be filled with such issues... Keep in mind if you're not seeing any interrupt activity you may have: 1. A problem with IRQs on your system Check the system for ACPI/APIC/IRQ/etc issues. You may need kernel parameters such as acpi=off, noapic, pci=assign-busses, etc... 2. A bad Sangoma card Their warranty is 5 years I believe, no better time than the present to request a new one under warranty. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Termination Provider Madness
- Original Message - Have a look at your /etc/asterisk/rtp.conf file. In it you specify the UDP portrange your asterisk will use for RTP traffic. change the rtpstart and rtpend to your needs and set them open in your FW. Do not make the range too small each active call will normally take one RTP channel incoming and one RTP channel outgoing. I have mine set to for example: rtpstart=1 and rtpend=10100. This should be enough for 100 simultanious calls. 2 RTP ports per session (inbound/outbound media)... that would mean 50 simultaneous calls, no? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer blocking CDR and recording?
- Original Message - No idea? ): How about showing your dialplan, and the log or console output from when you make the call? I have a hard time believing this number is special in any way... --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP DTMF Flash Event
Is there a way to have Asterisk respond appropriately when receiving a DTMF Flash event via SIP? I'm finding some WiFi SIP phones, specifically the Quickphones QA-342 want to handle transfers/3-way calls by sending a DTMF Flash event instead of handling it properly like every other damn VoIP phone on the planet... Asterisk sees the Flash event (via the logs), but does not act upon it. Thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.4.43 lost part of dialplan
- Original Message - On Thu, 20 Sep 2012, Jerry Geis wrote: Actually I restart asterisk every day at 2AM. So something happens in a 24hour window. On Thu, 20 Sep 2012, Jerry Geis wrote: THanks, actually all of my modifcations were to the extensions.conf file itself. It seems like those are the ones that got lost . Any chance you're running something like FOP that is updating your dialplan from a database? Jumping into the thread here, didn't see the opening post... Are you running FreePBX by any chance? It will overwrite your extensions.conf along with several others anytime you 'Apply Changes'. Maybe that's the case? Judging from 7+ years lurking on this list and never seeing anything like this, it's something weird about your install. Same here, something is amiss. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk
- Original Message - Yeah, I noted that too, but besides that it seems like it is exactly what I am looking for. I am especially confused that there's no hint like hey, buy our new product, just EOL. So let's say I am looking for an alternative to this. And unfortunately I have to add it's for private use and I therefore need a free solution, which probably restricts the selection ): Well, anything better than checking logs by hand would be already a good start :-) Sorry for digging up a zombie thread (Jun 20th or thereabouts)... I just stumbled upon Homer SIP Capture. It's 100% open source, and looks to be what you're in search of. Have a look: http://www.sipcapture.org/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy to install CDR-Viewer?
- Original Message - A simply PHP based thing would be OK. Maybe I should look more specifically for that or can anyone here recommend a PHP based CDR viewer? Meanwhile I ended up building a mysql view, for private purposes it does the job. A real solution would still be nice, though. Have a look here: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 This is for the old CDR stats package, which still works wonderfully well. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy to install CDR-Viewer?
- Original Message - just wondering if there is any easy to install CDR viewer? Easy meaning install some package (debian system) and that's it. Had some problems installing CDR-Stats, FreePBX also seems to be a longer task for setting up. Isn't there a simple (productive :p) solution? CDR-stat is about as easy as it gets, assuming you can setup a basic LAMP stack, and edit a config file or two (database parameters for CDRs). What issues are you having with that installation? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Easy to install CDR-Viewer?
- Original Message - - Original Message - just wondering if there is any easy to install CDR viewer? Easy meaning install some package (debian system) and that's it. Had some problems installing CDR-Stats, FreePBX also seems to be a longer task for setting up. Isn't there a simple (productive :p) solution? CDR-stat is about as easy as it gets, assuming you can setup a basic LAMP stack, and edit a config file or two (database parameters for CDRs). Caveat... I'm referring to the 'old' CDR-stats which was simple PHP based, not the 'new fangled' CDR-stats these young punks are using... :D --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
- Original Message -
Hello Ruben,
I belive the problem is not hylafax, is the way dahdi is configure,
here is
a part of the call log:
-- Accepting AUTHENTICATED call from xxx.xx.xx.xx:
requested format = ulaw,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing [xxx1463@fax-out:1] Dial(IAX2/503-2966,
dahdi/g3/xxx1463) in new stack
-- Called g3/xxx1463
-- DAHDI/4-1 answered IAX2/503-2966
As you can see above dahdi answered and IAXmodem thinks is the remote
fax
machine answered. That’s what I'm trying to change.
No, that simply means DAHDI successfully made the call out your POTS line, and
Asterisk has 'bridged' the call between your IAXmodem peer and the DAHDI
channel.
You could be experiencing some digit loss when dialing, causing the calls to
*sometimes* go through or not. Try adding a 'ww' to your dialstring to allow
the POTS line to settle before dialing:
Dial(DAHDI/g3/ww${EXTEN})
--Tim
--
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Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
- Original Message - Thanks Tim, I tried your suggestion below the logs: -- Accepting AUTHENTICATED call from xxx.xx.xx.xx: requested format = ulaw, requested prefs = (), actual format = ulaw, host prefs = (ulaw), priority = mine -- Executing [xxx1463@fax-out:1] Dial(IAX2/503-7761, dahdi/g3/wwxxx1463) in new stack -- Called g3/wwxxx1463 -- DAHDI/4-1 answered IAX2/503-7761 -- Registered IAX2 '503' (AUTHENTICATED) at xxx.xx.xx.xx:4570 [Aug 1 09:04:59] NOTICE[3392]: chan_iax2.c:8486 update_registry: Restricting registration for peer '503' to 300 seconds (requested 60) -- Registered IAX2 '503' (AUTHENTICATED) at xxx.xx.xx.xx:44145 [Aug 1 09:05:03] NOTICE[3391]: chan_iax2.c:8486 update_registry: Restricting registration for peer '503' to 300 seconds (requested 60) -- Hungup 'DAHDI/4-1' == Spawn extension (fax-out, xxx1463, 1) exited non-zero on 'IAX2/503-7761' -- Hungup 'IAX2/503-7761' [root@drew home]# faxstat -s HylaFAX scheduler on host.x.com: Running Modem ttyIAX0 (+1.xxx.8626): Running and idle JID Pri S Owner Number Pages Dials TTS Status 9126 S root xxx1463 0:1 1:12 16:10 No carrier detected Your setup looks correct. Can you connect a normal analog phone to the POTS line and dial that fax number directly? I just want to see if that is successful or not, indicating if the problem is PSTN related (need to dial 10 digits, or 1+10 for example in the US). The interesting thing is the result within Hylafax is 'No Carrier' which means the call was indeed answered, but fax was not present on the other side. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem
- Original Message - Yup, there is your problem. Tell hylafax to extend the amount of time before it times out. We're a bit off topic for the Asterisk list now, but in your Hylafax config.ttyIAX0 config file, add this: ModemWaitTimeCmd: ATS7=120 Restart Hylafax and faxgetty, then retry. That will allow 120 seconds on the dial before hanging up and assuming no carrier. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0
- Original Message - On 07/26/2012 03:32 PM, Danny Nicholas wrote: Question 1 - I think asterisk only supports a limited set of statuses Asterisk does not *receive* presence updates from Polycom phones (or really, non-Digium phones) at all. Instead, the presence (status) updates you are seeing appear on your phones are the statuses that Asterisk itself generates based on the phones' activity. Ah, I was suspecting that to be the case. Thanks for the info! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CAS T1 - No Ringback
Another mystery for the list, hopefully someone has ideas on a fix... :) I've got an Asterisk 1.8.12.0 system connected to a CAS T1 (ESF/B8ZS, fractional 1-8). Outbound dialing works correctly, but while the call is in progress, there is no 'ringing' heard by the end user. So, on a SIP phone connected to this system, I dial a number, that call goes out DAHDI via the CAS T1, and the remote side is actually ringing (my cell phone for example), but the SIP phone remains silent. If I answer my cell phone, full 2-way audio is present. The telco has already enabled ringback on the circuit but that has not had any effect on the operation. Any thoughts on how to proceed? Here are the pertinent parts of the debug log showing the events on the circuit when dialing: [Jul 27 11:31:20] VERBOSE[14199] app_dial.c: -- Called DAHDI/g1/XXX [Jul 27 11:31:20] DEBUG[14149] devicestate.c: No provider found, checking channel drivers for DAHDI - 1 [Jul 27 11:31:20] DEBUG[14149] devicestate.c: Changing state for DAHDI/1 - state 2 (In use) [Jul 27 11:31:20] DEBUG[14149] devicestate.c: device 'DAHDI/1' state '2' [Jul 27 11:31:20] DEBUG[14190] app_queue.c: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 27 11:31:20] DEBUG[14190] app_queue.c: Device 'DAHDI/1' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_WINKFLASH(3) on channel 1 (index 0) [Jul 27 11:31:21] DEBUG[14199] chan_dahdi.c: Channel 1: Sending 'T355885' to DAHDI_DIAL. [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Sent deferred digit string on channel 1: TXXXYYY [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:21] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_HOOKCOMPLETE(9) on channel 1 (index 0) [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:22] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_DIALCOMPLETE(6) on channel 1 (index 0) [Jul 27 11:31:22] DEBUG[14199] chan_dahdi.c: Enabled echo cancellation on channel 1 [Jul 27 11:31:22] DEBUG[14199] chan_dahdi.c: Engaged echo training on channel 1 [Jul 27 11:31:22] DEBUG[14199] chan_dahdi.c: Channel 1: Sending 'wwwYw' to DAHDI_DIAL. [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: analog_exception 1 [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: Exception on 19, channel 1 [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: __analog_handle_event 1 [Jul 27 11:31:24] DEBUG[14199] sig_analog.c: Got event ANALOG_EVENT_DIALCOMPLETE(6) on channel 1 (index 0) [Jul 27 11:31:24] DEBUG[14199] chan_dahdi.c: Echo cancellation already on [Jul 27 11:31:24] DEBUG[14149] devicestate.c: No provider found, checking channel drivers for DAHDI - 1 [Jul 27 11:31:24] DEBUG[14149] devicestate.c: Changing state for DAHDI/1 - state 6 (Ringing) [Jul 27 11:31:24] DEBUG[14149] devicestate.c: device 'DAHDI/1' state '6' The odd part is, you can see above the dialed number was XXX, but the actual sequence on the trunk as performed was to dial XXXYYY, then some 'waits', then the last digit Y. Is this normal? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom Presence with Asterisk 1.8.12.0
Greetings- I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0 system. Everything is running smoothly with few problems. However, I have an issue that maybe someone could shed light on... Many of the phones have 'buddy watch' enabled for the other phones, basically Polycom's version of BLF. This works fine when watched extensions are on the phone, ringing, etc, as the LED lights/flashes appropriately for the status. However, the phones also offer various presence states such as 'Out to Lunch' or 'Away from Desk' etc. When a phone is set to one of these presence states, the other phones watching never show that status. Does that make sense? Is there any reason why those states would not propagate between the phones (through Asterisk?) ? And, on a side note, if anyone knows how to remove the 'thistle' background from a Polycom phone I'd be especially delighted. It was set by a user on a device, and there is no option to remove it, or replace it with the blank background which is the default. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
- Original Message - Thanks Tim. One of the problem that I am facing is the complicated generated configuration for the FreePBX, is it the same thing in the Elastix? To understand this complicated generated commands, is there a documentation to explain this for FreePBX or Elastix? One of my friend told me that he installed (as I remember) FreePBX and there were already existed the TFTP files that the Cisco IP Phones is requesting (for sip or skiny) and already there were a TFTP server. Which module to do this? FreePBX (and any other project based on it, including Trixbox, Elastix, PIAF, AsteriskNow, etc) stores all information in a MySQL database, then when you click 'apply changes' it takes all of the system config info from the database, and generates the Asterisk dialplan code in /etc/asterisk . This is the tradeoff for having a magical GUI do most of the work, and doing everything by hand. You can of course make your own changes that will not be overwritten by editing the appropriate *_custom.conf files. For example, to add contexts and/or dialplan stuff, put it in extensions_custom.conf. The same applies for sip_custom.conf, etc... Most of the predone projects (Elastix is my favorite at the moment) include some sort of endpoint manager that will generate configs for your phones. I'm not sure specifically on Cisco phones, other than they are a huge PITA in general. The system just needs a TFTP server installed, and the phones pointed to it (manually, or by using DHCP option 66). --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
- Original Message - I'm currently trying to decide on which GUI-enabled version of Asterisk to use for one particular installation, where we will need good telecommuter support. We've made it so easy for people to work remotely that the customer is downsizing their real estate and will have 90% remote workers with them rotating through the office as needed. So most phones in the office will be shared, and I'm looking for a version of Asterisk that will easily allow people to log in and out of a specific desk. What are your suggestions? I have very little experience with GUI versions of Asterisk; we use bare Asterisk for nearly everything. Not to sound like a broken record or anything... but I'd say give Elastix a go. It is top notch in terms of release quality and features. And, being based on FreePBX, you can set it to 'Device and User' mode instead of the default extensions mode so users can 'hotdesk' between phones. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I install Asterisk normally and then installing the GUI asterisk now
- Original Message - Hello; Is it possible if I have already asterisk installed on Fedora machine to install the GUI asterisk now without doing a fresh installation using the Asterisk Now CD? Which version of the GUI that should be selected to work with the asterisk version? For example, if I have asterisk 1.8 then which GUI version to select? I am talking about compatibility. Can I say that Freepbx is Asterisk + Asterisk Now? You could, but it would be wrong. :) AsteriskNow is Asterisk+FreePBX in a nutshell. Of course there is better package management (RPM repos from Digium vs source installs or building your own packages), etc. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI DTMF problem?
- Original Message - It has a Digium Wildcard TE122 If it has an onboard echo canceler, try disabling it and retrying. Just a shot in the dark, going from my experience with other cards and same symptoms. If the card is new(ish) I would think Digium could provide support to you for determining the DTMF problems. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I install Asterisk normally and then installing the GUI
- Original Message - OK, is there asterisk-gui that differs that freepbx? Or Freepbx is a GUI for asterisk? In other words, if I have asterisk and I need to add for it a GUI, is there asterisk-gui which is differs than freepbx or it is the same? There have been a handful of other GUIs around, but none have the market share, support, or feature-set of FreePBX. It is pretty much the defacto standard. If you're feeling adventurous, give asterisk-gui a try. Last I checked, the latest code was available from Digium SVN and 'worked better' than from the tar.gz on downloads.digium.com. But again, that was some time ago, YMMV... --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX?
- Original Message - Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Trixbox includes FreePBX as it's GUI. However, keep in mind it is a bastardized, forked version of FreePBX that has seen nary any new development or innovation in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX?
- Original Message - Hi Tim, How about AsteriskNow? Thanks and BR, Anam On 7/7/12, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Hi All; Based on what I have to use Trixbox or FreePBX? Can someone advise? Trixbox includes FreePBX as it's GUI. However, keep in mind it is a bastardized, forked version of FreePBX that has seen nary any new development or innovation in some time. At this point, for a standard PBX installation, my recommendations would be (in this order): 1. Elastix 3. AsteriskNOW 2. PBX In a Flash Did you read #2 above? Erm, wait, #3 I guess? The list is in proper order but apparently my ability to make a numbered list is somewhat lacking today. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip and extensions
- Original Message - I am new. Here is the code that I am playing with on CentOS 6.x register = 5552530146:funnytiger...@sip3.voipvoip.com [outgoing] username=5552530146 type=peer qualify=yes secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromuser=5552530146 fromdomain=69.90.209.57 dtmfmode=rfc2833 allow=g729 allow=ilbc allow=ulaw allow=alaw disallow=all srvlookup=no [incoming] username=5552530146 type=user secret=funnytiger123 nat=auto insecure=very host=69.90.209.57 fromdomain=69.90.209.57 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc disallow=all srvlookup=no *PLEASE* if that is your real username/password with your VoIP provider change it immediately. Just FYI, you've broadcast it to (tens or hundreds of) thousands of list readers. I have to believe some are of the nefarious type that would love to use your account for free calling at your expense. :/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
- Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI trunk between Asterisk servers does not work.
- Original Message - Quoting Tim Nelson tnel...@rockbochs.com: - Original Message - Curiously enough, I can't do that at all on Voip3. Not span 3 of course, because only span 1 should exist, but I can't execute pri show spans either. DING DING DING... we may have a winner. Do you have PRI support on that box, meaning, did you also compile libpri before compiling Asterisk? How about watching your Asterisk log files during Asterisk startup to see any output of when chan_dahdi.conf loads? (tail -F /var/log/asterisk/full) Excellent! Funny thing about that. Our original plan was to use a SIP trunk until we discovered that faxes don't work worth a damn that way. Ergo, I didn't compile libpri first. Yep, that'd cause what you're seeing. Glad we could help. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] .lock file issue
- Original Message - I'm currently running Asterisk 10.5.1, compiled from source, and just had someone call saying they couldn't get their voice mail. Looking into the user's voice mail folder, I saw a .lock file. Removing this file, enabled them to get voice mail. Is anybody else seeing this? The system is a new install and has only been running for a week with very little traffic (8 person office). This was quite common in some old releases, at least for me. At one point I wrote a quick script that ran via cron to remove those lock files once per minute. If this isn't a new bug, it could also be a full filesystem, or maybe the system lost power during an event where a lock was created but not removed? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium IP Phones D40
- Original Message - We have the ringer volume issue with some customer environments as well. We use Grandstream phones in a lot of installs so we just upload a custom ringtone with the db pushed up on it a bit. We are testing the Digium phones and have concerns if we will be able to use them for the high noise env customers. Polycom phones do have the same ring volume issue for these customers. No issues in general office env. For everyone complaining about the Polycom's lack of volume, are you simply hitting the volume buttons, or are you also aware of the myriad of adjustments available in the Polycom XML provisioning configs? We have a local educational customer that experienced volume problems with Polycom due to noisy classroom environments, and with a few tweaks to volume and gain in the XML configs pushed via TFTP, the phones were ear-splittingly loud, both ringers and handset. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk
- Original Message - Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems. 2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data? So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality. Have a look at VQmonitor: http://www.manageengine.com/products/vqmanager/ It works very well. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Proactive problem monitoring on SIP on Asterisk
- Original Message - - Original Message - Hello, 1) I am wondering what is the best practice to monitor if there are or were problems with SIP calls on my Asterisk box. E.g. how about a software that extracts all calls from the /var/log/asterisk/full (I have permanently enabled verbose 10 and sip debug) log and tells me on which of them were problems? Checking the logs manually is very hard, but as SIP is a standardized protocoll, there should be tools doing that for you? As an example, a person calling me recently got a 488 Not acceptable error as reply from my Asterisk box. Nothing came through to my SIP phone, so I didn't know anything about the call or the problems (which were on his phone btw). I would like to be informed about such cases, know that there was a call to my Asterisk box that made problems. 2) How about monitoring speech quality? E.g. sometimes it seems like a packet is missing (I then have a short pause during the call), how to monitor such things and create statistics out of this data? So basically I want to monitor my Asterisk installation proactively for reliability/problems and (speech) quality. Have a look at VQmonitor: http://www.manageengine.com/products/vqmanager/ It works very well. ...it worked well when you could buy it. Apparently it is EOL now [1]. Sorry for the noise. These aren't the droids you're looking for. --Tim [1] http://www.manageengine.com/products/vqmanager/eol.html -- _ -- 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] SCCP Questions
Greetings Ron- Just wanted to give you a heads up about an alternative SCCP channel driver available for Asterisk. Please see here: http://freecode.com/projects/chan-sccp-b I have no experience with it (nor SCCP in general) but just wanted to give you an option in the event the included SCCP driver does not give you satisfactory results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax over IP?
- Original Message - Hi Tim, I'm using Asterisk 10 and on Cisco GW the protocol is set for FAX is T.38 and when I try to send the fax from a fax machine i.e. HP 3180, I'm getting some warnings as listed below; -- Executing [4112345678@default:1] Goto(SIP/192.168.1.69-0005, fax-detect,fax,1) in new stack -- Goto (fax-detect,fax,1) -- Executing [fax@fax-detect:1] NoOp(SIP/192.168.1.69-0005, FAX DETECTED ) in new stack -- Executing [fax@fax-detect:2] Goto(SIP/192.168.1.69-0005, fax-receive,receive,1) in new stack -- Goto (fax-receive,receive,1) -- Executing [receive@fax-receive:1] NoOp(SIP/192.168.1.69-0005, FAX RECEIVE ) in new stack -- Executing [receive@fax-receive:2] Set(SIP/192.168.1.69-0005, GLOBAL(FAXCOUNT)=5) in new stack == Setting global variable 'FAXCOUNT' to '5' -- Executing [receive@fax-receive:3] Set(SIP/192.168.1.69-0005, FAXCOUNT=5) in new stack -- Executing [receive@fax-receive:4] Set(SIP/192.168.1.69-0005, FAXFILE=fax-5-rx.tif) in new stack -- Executing [receive@fax-receive:5] Set(SIP/192.168.1.69-0005, GLOBAL(LASTFAXCALLERNUM)=6461234567) in new stack == Setting global variable 'LASTFAXCALLERNUM' to '6461234567' -- Executing [receive@fax-receive:6] Set(SIP/192.168.1.69-0005, GLOBAL(LASTFAXCALLERNAME)=) in new stack == Setting global variable 'LASTFAXCALLERNAME' to '' -- Executing [receive@fax-receive:7] NoOp(SIP/192.168.1.69-0005, SETTING FAXOPT ) in new stack -- Executing [receive@fax-receive:8] Set(SIP/192.168.1.69-0005, FAXOPT(ecm)=yes) in new stack -- Executing [receive@fax-receive:9] Set(SIP/192.168.1.69-0005, FAXOPT(headerinfo)=MY FAXBACK RX) in new stack -- Executing [receive@fax-receive:10] Set(SIP/192.168.1.69-0005, FAXOPT(localstationid)=1234567890) in new stack -- Executing [receive@fax-receive:11] Set(SIP/192.168.1.69-0005, FAXOPT(maxrate)=14400) in new stack -- Executing [receive@fax-receive:12] Set(SIP/192.168.1.69-0005, FAXOPT(minrate)=2400) in new stack -- Executing [receive@fax-receive:13] NoOp(SIP/192.168.1.69-0005, FAXOPT(ecm) : yes) in new stack -- Executing [receive@fax-receive:14] NoOp(SIP/192.168.1.69-0005, FAXOPT(headerinfo) : MY FAXBACK RX) in new stack -- Executing [receive@fax-receive:15] NoOp(SIP/192.168.1.69-0005, FAXOPT(localstationid) : 1234567890) in new stack -- Executing [receive@fax-receive:16] NoOp(SIP/192.168.1.69-0005, FAXOPT(maxrate) : 14400) in new stack -- Executing [receive@fax-receive:17] NoOp(SIP/192.168.1.69-0005, FAXOPT(minrate) : 2400) in new stack -- Executing [receive@fax-receive:18] NoOp(SIP/192.168.1.69-0005, RECEIVING FAX : fax-5-rx.tif ) in new stack -- Executing [receive@fax-receive:19] ReceiveFAX(SIP/192.168.1.69-0005, /var/spool/asterisk/fax/fax-5-rx.tif) in new stack -- Channel 'SIP/192.168.1.69-0005' receiving FAX '/var/spool/asterisk/fax/fax-5-rx.tif' == Using UDPTL CoS mark 5 [Jun 4 12:35:02] NOTICE[10371]: chan_sip.c:7577 sip_read: FAX CNG detected but no fax extension [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1666 receivefax_t38_init: channel 'SIP/192.168.1.69-0005' refused to negotiate T.38 [Jun 4 12:35:02] WARNING[10072]: res_fax.c:1687 receivefax_t38_init: Audio FAX not allowed on channel 'SIP/192.168.1.69-0005' and T.38 negotiation failed; aborting. [Jun 4 12:35:02] ERROR[10072]: res_fax.c:1891 receivefax_exec: error initializing channel 'SIP/192.168.1.69-0005' in T.38 mode == Spawn extension (fax-receive, receive, 19) exited non-zero on 'SIP/192.168.1.69-0005' In my sip.conf global configuration I enabled 'fax detect' and 't38pt_udptl' and added Cisco VGW peer; [CiscoVGW-10.70.X.X] host=10.70.X.X type=friend disallow=all allow=ulaw allow=alaw nat=yes insecure=port,invite context=fax-call canreinvite=no qualify=yes dtmfmode=inband T.38 failed to negotiate. That means either your Asterisk side, or your Cisco side are not playing nicely together. A packet capture of the call setup would be helpful to determine which side is having the issues. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax over IP?
- Original Message - Hi Tim, ... While the fax machine starts to send the fax after a while it gives the message, 'Fax failed' with error code: '388'. Is it the end point fax machine issue or else? Please assist me out to resolve this issue at earliest. Please do not email me directly. I've already responded on list, despite wanting to let this sit for a few days in response to you asking for support 'at earliest'... The Asterisk support list has no SLA, only governed by the time and willingness of the members to participate. Thanks. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma D100 Transcoder Asterisk 1.6
- Original Message - I have installed and configures this card in asterisk 1.6. When trying to load the module codec_sangoma.so I see the following in the asterisk log. [2012-06-04 15:50:31] WARNING[18168] loader.c: Error loading module 'codec_sangoma.so': /usr/lib/asterisk/modules/codec_sangoma.so: undefined symbol: ast_config_load [2012-06-04 15:50:31] WARNING[18168] loader.c: Module 'codec_sangoma.so' could not be loaded. Has anyone had a similar issue with this card or have any idea what the undefined symbol: ast_config_load might mean to figure out what direction to head for further debugging? It looks like maybe Wanpipe was not compiled against the same version of Asterisk/DAHDI you're running. That would be the first thing to check. Next stop, Sangoma support. They are fantastic, and support is free. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax over IP ?
- Original Message - Hi all, Couple of things I would like ask, does Asterisk provides free license for FoIP (for 1 channel) or need to purchase it? Couple of years back, I was able to send and receive the fax using Digium T1 card, in term of FoIP how can I able to receive fax from traditional telephone lines / T1 lines? As far my understanding, the functionality for FoIP is to send fax to email or receive fax from email i.e. using T.38 protocol. The thing I would like to know how I can implement this solution i.e. receiving fax via IP? Correct me if I'm wrong, while receiving fax from traditional telephone lines will the topology looks like as listed below; PSTN Lines -- Asterisk (mounted a T1/ analog card) -- IP -- Asterisk (receive Fax over IP) or else? FoIP typically means the fax session traverses an IP link at some point, most commonly at the 'last mile'. What happens to the fax after that is up to your requirements. The faxes can be emailed out, stored in a web application, printed to a printer, etc. The possibilities are endless. Asterisk does have a few options for faxing. Those are most notably: 1. Fax for Asterisk - Free license available for 1 channel, or paid licenses for 2+ channels 2. app_fax (I think this is the current module name) - Free fax module for Asterisk, no channel limit, based on SpanDSP 3. Hylafax+ and IAXmodem - Most complicated method of fax setup, but most robust and reliable (in my testing). Would require use of Asterisk 10 with T.38 gateway functionality for proper fax reception. Just keep in mind raw fax audio over VoIP is a bad idea, see here: http://www.soft-switch.org/foip.html If you can provide some additional details on what you're planning to do, we can give more info. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Server for Asterisk
- Original Message - Hi Tim, Unfortunately i can't reproduce the scenario because it was a long time ago. But it would be nice to hear from you, what things can be verified within fax and Asterisk? Any TIP on wireshark monitoring? Within Asterisk, the debug logs can be helpful for routing/connectivity diagnostics. With Hylafax, all of your details will be found in the session logs in /var/spool/hylafax/log. Here you can see each session's interaction with the remote fax device. It is an art deciphering the various protocols, but the folks on the Hylafax lists are incredibly helpful until you've learned the magic of understanding the logs directly. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax over IP?
- Original Message - Hi Tim, Thanks for your response. Here is my topology as listing down below; PSTN Line -- Cisco Voice GW -- IP Cloud -- Asterisk Will Asterisk able to receive the fax (as in topology above) using its' fax module? In sip.conf I enabled fax detection and T.38. Actually I don't want to use Hylafax + iaxmodem as per requirement. If your Cisco voice gateway can deliver the calls using T.38, that should give you decent reliability. You'll want to us Asterisk 10 which has the best T.38 support at this point (compared to older releases). The receiving side of the equation then becomes whether to use Fax for Asterisk (commercial, 1 free channel, 2+ paid), or the included SpanDSP based fax module. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Server for Asterisk
- Original Message - I've used Asterisk 1.4.22 with hylafax and iaxmodem and it wasn't reliable at all; sometimes the fax reach the destination, sometimes not, and even worse, asterisk got froozen...(here using analog lines over Sangoma B600 and Digium TDM400P, same behavior with both. Other history with same asterisk version but E1 lines, it was PERFECT. That's why i ask for analog lines, since not all customers has E1. Any recommendation/restriction when using hylafax + Asterisk + iaxmodem ? BR If Asterisk was freezing up, that would seem to indicate a problem with Asterisk, not the Hylafax/IAXmodem components. Of course, details would be needed to determine why that was the case. Regardless, without lockups of Asterisk, reliability of fax is very dependent on timing and audio quality. Again, details would be needed to further investigate why you had high failure metrics(specifically your fax session logs from /var/spool/hylafax/log). In general, Hylafax+[1] and IAXmodem is the most rock solid stable fax solution available, as long as you can get past the initial learning curve. There is a reason why IAXmodem has not had a release in forever as the 1.2.0 release is rock solid stable. Hylafax+ continues to be developed with regular releases, the feature set and functionality are second to none with hooks for almost any imaginable configuration, and the support via the mailing lists or available contractors can't be beat. /soapbox If you have specifics about your problems with Hylafax and IAXmodem, I'd love to hear about them to help diagnose, if it is postmortem. --Tim [1] There *IS* a difference between Hylafax (hylafax.org) and Hylafax+ (hylafax.sourceforge.net). Please see here: http://hylafax.sourceforge.net/about.php -- _ -- 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] Fax Server for Asterisk
- Original Message - On Tue, May 29, 2012 at 3:10 AM, Danny Dias ing.diasda...@gmail.com wrote: Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks I've been real happy with using HylaFax+ and Iaxmodem implementations. +1 for this recommendation. Integrated approaches such as app_fax/res_fax may be 'easier', but you'll never the amount of customization, tunability, and control available with IAXmodem and Hylafax+. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting Fax Tones over IAX2
- Original Message - On 05/23/2012 08:41 PM, Cody Harris wrote: Hello All, I use IAX2 as the incoming connection from my DID provider. For whatever reason, this works best for me, SIP connections lag very frequently and only have about a 50% success rate for incoming calls (they get dropped mysteriously). I'm trying to implement a fax/voice switch. I have faxdetect=both in my sip.conf, and when I use sip, it works well. However, from what I can tell, there's no such option for IAX2 connections. Any ideas on what I can do here, or am I out of luck? It's quite hard to provide suggestions since we don't know what version of Asterisk you are using. However, in Asterisk 10, there is a channel-agnostic FAX detection function that can be applied to any channel type, so at a minimum that is one way to solve your problem. BUT, even if fax is detected on an IAX2 channel, the only reason would be to change dialplan logic accordingly correct? There is no T.38 equivalent within IAX2, which means the OP will be handling faxes over a clear VoIP channel. The information here is of utmost relevance: http://hylafax.sourceforge.net/docs/fax-over-voip.pdf --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
- Original Message - Hi Steve, you are telling me there is no way to set a particular speed on my iaxmodem in order to force the sender speed? I have some problems with a customer who gets malformed faxes even if no error occurs. Since I cannot tell the sender to lower its fax speed, my idea is to force my iaxmodem to a lower fixed speed so the sender is oblidged to negotiate at that speed (or lower, of course) without the customer could realize it, at least at first. :) There is no ATA in the middle (I'm using it for my tests but my customer does not have any), all faxes are received thru a primary channel to a bunch of iaxmodems. Sometimes some faxes are corrupted, that's why I thought to lower the speed. I could try to disable ECM but that's even harder to do (found nothing on internet). You're getting corrupted fax data and want to solve that problem by *disabling* ECM? That seems counter-intuitive to me... How are your fax calls coming into your system (PSTN-???-Asterisk-IAXmodem-Hylafax)? If you have VoIP somewhere in the call path, you'll likely keep bashing your head on the table trying to fix problems that will never go away. Also, don't be afraid to recognize sometimes your side (as the receiver) is working perfectly well, and sometimes there just isn't anything you can do about senders on bad lines/sending over VoIP/etc. The quality of a fax session is only as good as the weakest link contained within that session, including the call path from sender to receiver. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
- Original Message - Hi guys, thanks for answers. That could seem counter-intuitive but it is not. Not to mention the fact that information technology is not science, Huh? It is indeed very much a science. You have known established facts, processes, concepts, methods for testing and implementing those elements. the solution to broken faxes is to lower down speed. This works even with normal telco lines even *Sometimes* lowering speeds can help get faxes through, but you're missing the big picture. Fax is sensitive to latency, jitter, timing, interference, audio clipping, etc. Simply lowering your speed will not magically make all of those issues go away. You need to be looking at finding a solution to the root cause of the problem, not throwing some odd ideas at the symptoms. if you DO NOT have a pbx (telco technicians even say not to make faxes pass thru your PBX). I could ask my customer's telco to lower the speed down How would the telco lower the speed of your customer's fax gear? Unless they have direct control of it, or are providing some sort of T.38-T.30 service, the only way I can think of is they would have to introduce a problem on the line that would force the fax sender/receiver to force a lower speed during normal negotiation. Someone please correct me if I'm wrong? but it depends on the guy working at the call-center...sometime you talk to dummy people who ARE sure it is impossible. But it is not. So, I do not want to spend days to convince people working at that telco call-center that what I'm asking is feasible and I do not want to tell my customer to tell their customer to lower their faxes speed (before installing our PBX they were able to send perfect faxes so, why should they?). Again, your PBX/equipment/whatever can be 100% the best, most reliable system ever to be installed. BUT, if you have senders that are on poor lines, running over VoIP, or have a multitude of other issues, the problem lies with them. There is not much you can do to solve this. My idea was to tell iaxmodem not to accept fast speed rates so the fax machine on the other side should be forced to negotiate a slower speed as if my customer fax weren't virtual as iaxmodem is but a real one. I suspect that the problem is about the primary lines because I tested iaxmodem many times on my LAN and it is (surprisingly :) ) working fine Yes, performance *can* be good on an unloaded LAN. But again, it is fax over VoIP which means tomorrow it may not work because Jim Bob over in accounting is updating Windows, watching Youtube, downloading some music via Bittorrent, and backing up his machine to the fileserver. Point being, network performance is 100% responsible for your local IAXmodem experience over the LAN. :) (10 good received faxes out of 10 sent!!!) but, as you may know, talking to telco technician is a nightmarethey always say problems are always on the PBX side... :( I'm sure that is standard procedure. If you were in their shoes, would you want to deal with every possible PBX issue that comes around? I'm not saying it's right, just that's the way it is. Moreover, after sending a fax, the fax machine beeps correctly as the fax was correctly sent without corruption. :o No, the fax machine beeps to say the fax was *SENT*. Whether or not there was any corruption is entirely up to the sender/receiver to determine, typically with copy quality checking or ECM. I hope I have made my point but I'll try do dig deeper inside the problem as you suggested me. A point was indeed made. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to set iaxmodem receiving speed
- Original Message - On 05/17/2012 07:53 AM, Andrew Furey wrote: we use ActiveFax for sending (interfaced from an ERP package) and often get Comm Error 283 and incomplete faxes. If it's just making a bad situation worse, how is it that our solution of turning off ECM mode fixes it 98% of the time? I'm curious. Because apparently the ECM protocol in ActiveFax is broken. If disabling a feature that is designed to *improve* fax reliability and performance actually does the opposite, then there's no other explanation than to conclude that the implementation of that feature is broken in the product you're using. I was about to write a response to this, you nailed it on the head. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
- Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Have a look at the Blackbochs SBC***. It is small, low power, plenty of storage options, built in analog telephony ports, etc: http://www.rockbochs.com/products/blackbochs-sbc --Tim ***Yes, I'm affiliated with the product/company, but it is on topic for this discussion. My apologies if this offends anyone. -- _ -- 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] looking for solid state like PC suitable for Asterisk
- Original Message - On Thursday 10 May 2012, Bart Coninckx wrote: I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Does anyone have inspiration/experience for/about such a model? Raspberry Pi would be the obvious choice, surely? The hype around the Raspberry Pi is enormous. I would not consider it a real option for production voice until it's had a chance to mature and be available for some time to iron out the bugs, both hardware and software related. My $0.02 USD. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
- Original Message - Tim, looked at these briefly, they all seemed pre-installed, correct? Is reinstallation with, let's say, CentOS possible? thx, BC The units *can* come preinstalled with our PBX flavor (Debian, Asterisk, FreePBX), or they can be sent bare and you can install your OS/platform of choice. CentOS specifically does not run on the board as the upstream vendor does not support i586 arch any longer (since Centos 5.x series IIRC). We've done some work trying to patch the installer and use custom kernels to get around this, but were unsuccessful. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] looking for solid state like PC suitable for Asterisk
- Original Message - On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) I think most users are just trying to be specific about not wanting any computer equipment where tubes[1] are in use. :D --Tim (...who still uses and loves his tube audio gear...) [1] http://en.wikipedia.org/wiki/Vacuum_tube -- _ -- 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] looking for solid state like PC suitable for Asterisk
- Original Message - On 05/10/12 18:38, Kevin P. Fleming wrote: On 05/10/2012 03:49 AM, Bart Coninckx wrote: Hi all, for smaller (or maybe even bigger) sites I'm looking for a smaller, appliance-type like PC, preferably solid state and fanless PC. Since it's only going to run Asterisk for a couple of extensions I don't think CPU and RAM need to be maxed out. Just a small comment here... I really find it quite humorous that people use 'solid state' to mean 'no moving parts'. All of the parts of my computers that move are still composed of solid materials, and the electrical currents involved in them still move through solid materials :-) Yeah, well, have you seen crawling any bugs in software lately? Still they are called bugs ... :-s Funny, I've heard them referred to as 'features'. :D --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DAHDI FXO Call Issues / Indication Types
Greetings- I've had reports of a customer PBX acting strangely to some inbound calls. Specifically, a call comes into an FXO port, hits a Dial() to ring a few extensions, but by the time someone answers the phone, the call has been dropped, and the caller is listening to on-hold music. There is nothing in the dialplan that would cause this behavior. The logs have a couple of odd (to me, maybe normal) items that may give some information. After the call comes in, and the SIP endpoints are dialed, I see: [2012-04-12 08:51:04] DEBUG[7313] chan_dahdi.c: Requested indication 3 on channel DAHDI/1-1 I have to assume indication 3 means playing ringing tones to DAHDI/1? The phones are ringing at this point. In the logs I see this: [2012-04-12 08:51:06] DEBUG[7313] sig_analog.c: analog_exception 1 [2012-04-12 08:51:06] DEBUG[7313] sig_analog.c: Exception on 13, channel 1 [2012-04-12 08:51:06] DEBUG[7313] sig_analog.c: __analog_handle_event 1 [2012-04-12 08:51:06] DEBUG[7313] sig_analog.c: Got event ANALOG_EVENT_RINGBEGIN(12) on channel 1 (index 0) I'm assuming the indication 3 specified earlier was to send ring tones on DAHDI/1, this simply confirms it? Next, one of the phones answered the call, and a new indication is sent: [2012-04-12 08:51:07] VERBOSE[7313] app_dial.c: -- SIP/102-21ae answered DAHDI/1-1 [2012-04-12 08:51:07] DEBUG[7313] chan_dahdi.c: Requested indication 22 on channel DAHDI/1-1 What does indication 22... indicate? There is more to the log, but it is intermingled with a callback and I'm not easily able to parse it out at the moment. Does any of the above seem out of place, or does it look 'normal'? As an aside, where could a person find a list of indications used on DAHDI channels, analog and/or PRI/T1? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Slightly OT] Audiocodes Mediant Failover Routing with Asterisk
- Original Message - Greetings- First off, my apologies for the slightly OT nature of this post. It does involve Asterisk to a degree, but errs a bit on the side of Audiocodes inquiry. I accept all responsibility for my actions and the consequences. :) The scenario is this: I have an Asterisk box connected to a Mediant 2000 (M2K) via T1. Calls made via DAHDI-T1-M2K are then routed by the M2K via SIP using the Audiocodes Tel-to-IP routing tables to a remote IP. This has worked quite well for some time. I now want to add multiple IP's to the routing table for failover. However, when adding additional entries to the Tel-to-IP routing table, and the first entry fails, the other IPs are not attempted. The first IP is attempted, and if it fails, the call fails instead of trying any of the additional IPs in the Tel-to-IP routing table. *BUMP* Any ideas from the brilliant folks here? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Slightly OT] Audiocodes Mediant Failover Routing with Asterisk
Greetings- First off, my apologies for the slightly OT nature of this post. It does involve Asterisk to a degree, but errs a bit on the side of Audiocodes inquiry. I accept all responsibility for my actions and the consequences. :) The scenario is this: I have an Asterisk box connected to a Mediant 2000 (M2K) via T1. Calls made via DAHDI-T1-M2K are then routed by the M2K via SIP using the Audiocodes Tel-to-IP routing tables to a remote IP. This has worked quite well for some time. I now want to add multiple IP's to the routing table for failover. However, when adding additional entries to the Tel-to-IP routing table, and the first entry fails, the other IPs are not attempted. The first IP is attempted, and if it fails, the call fails instead of trying any of the additional IPs in the Tel-to-IP routing table. Any thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - T.38 unreliable on a LAN : truth or obscurantism ?
- Original Message - Hi, When someone says T.38 is not reliable on a (normally loaded and managed) LAN, would you rather agree or disagree ? In this case, fax calls are coming in through an analog gateway, passing trough Asterisk and then going out to ISDN through a digital gateway. Is T.38 actually in use in this scenario? Or are you simply passing the fax call through Asterisk as 'normal' audio (G.711u/a, etc)? If so, you may want to see here: http://www.soft-switch.org/foip.html --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a php script to analyse and show call detail reports from Asterisk CDR?
- Original Message - Yes, this is exactly what I am looking for - hopefully in English :-) Date or range selection would make this perfect. I have been looking for something like this for quite a while but there is none. I would really appreciate it if you share this with me. Question here, does the .php code read from database and displays or does it analyse the custom-cdr.csv file? Don't forget about the ever-popular Asterisk-stat and the newly revised cdr-stats projects, both web based, proven, and work fantastic: http://www.areski.net/areski/index.php?option=com_contenttask=viewid=22Itemid=54 http://www.cdr-stats.org/ --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question for the group
- Original Message - Hello Folks; I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions We are going to be setting up a prepaid PBX service with the following features: • Email to Fax and Fax to Email • Inward DID local and 800 services • Calling card SIP based and ANI authenticated I see there are many types of software that can be addons/installs/etc to Asterisk. So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does) So, any thoughts? You just posted this to the asterisk-biz list under a different name/email address. The one response you received was immediately brushed off because you apparently cannot read: Thanks for this - but I am looking really for a software type solution. The product offered *IS A SOFTWARE SOLUTION* that would run on your hardware. The posted option is more than suitable to your needs, and offered by folks with a highly deserved great reputation. Good luck to you. --tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality
Greetings- I currently have a customer that *requires* key-system functionality in an Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of the analog lines attached to the system (DAHDI FXO). By pressing one of these keys (for line 1 for example), the dialed number needs to be dialed out the correct port. Also, when that line is busy, the phone BLF key for that line needs to reflect the status. I've been reading about SLATrunk but it doesn't seem quite what I'm looking for. Also, I'm looking at using Hints to supply such information, but I'm not sure exactly how it should look. Has anyone done this before, and if so, how did you implement it? My target is to use Asterisk 1.8 but another version would suffice. Looking forward to your comments! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SLA for DAHDI FXO - Emulating Key System Functionality
- Original Message - On 01/26/2012 09:46 AM, Tim Nelson wrote: Greetings- I currently have a customer that *requires* key-system functionality in an Asterisk PBX. On a SIP phone, the BLF keys need to show the current state of the analog lines attached to the system (DAHDI FXO). By pressing one of these keys (for line 1 for example), the dialed number needs to be dialed out the correct port. Also, when that line is busy, the phone BLF key for that line needs to reflect the status. I've been reading about SLATrunk but it doesn't seem quite what I'm looking for. Also, I'm looking at using Hints to supply such information, but I'm not sure exactly how it should look. Has anyone done this before, and if so, how did you implement it? My target is to use Asterisk 1.8 but another version would suffice. The SLA functionality is exactly what you are looking for. Fantastic, I'll revisit the usage and do some testing. However, can anyone point me to some working examples? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which SpanDSP version to play with Asterisk 10 and T.38/T.30 gatewaying ?
- Original Message - I use the latest spandsp source from the freeswitch git. There you have also a changelog documenting the differences. Steve Underwood commit here the latest changes in spandsp source. http://fisheye.freeswitch.org/changelog/freeswitch.git/libs/spandsp Does this represent the latest developments in SpanDSP code? Or, is this just a more publicly available location for the source (in progress) as soft-switch.org only appears to have specific releases? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 10.0 1.4 - iax codec are not compatible
- Original Message - I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have 11, digest has pstn-1270 NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device KMIEC Z sip:7804715665@10.0.0.110;tag=1c1222950155 Anybody know what is the magic solution to get a CallerID to work? In addition iax -codec are not compatible with earlier asterisk (1.4); I have selected ulaw / alaw but Asterisk 1.4 wants GSM: chan_iax2.c:9541 socket_process: Rejected connect attempt from 192.168.141.8, requested/capability 0x2/0x703 incompatible with our capability 0xc. IAX is a signalling protocol, not a codec. And, interop between the various branches of Asterisk is not a problem. Looking at your logs, the problem appears to be you have a codec mismatch between peers, or your authentication details are wrong. Please also check that you have calltokens set the same on both sides (enabled on both or disabled on both). --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to stop hacking of my server
- Original Message - On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com wrote: Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. Odd nobody else mentioned it yet, so I'll do it... Check out fail2ban. If you have peers or systems that you cannot restrict by IP and must leave relatively 'open', fail2ban will see the failed attempts, and after a configurable number of failures, will automatically add the offending IP to IPtables. See here: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to stop hacking of my server
- Original Message - Le 27/12/2011 16:04, Tim Nelson a écrit : - Original Message - On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com wrote: Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? [...] Odd nobody else mentioned it yet, so I'll do it... Check out fail2ban. [...] He said except iptables. fail2ban is iptables related ;-) Ahhh, yes, it would probably have helped if I read the message in it's entirety. :) --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Number of Calls
- Original Message - Hi, I am new in voip, how many calls can one asterisk box handle with 30 % of trans-coded calls and system configuration as 8GB RAM X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo This is one of the 'harder' things to calculate. You'll at least want to start here for some ideas: http://www.voip-info.org/wiki/view/Asterisk+dimensioning In general however, you appear to have some beefy hardware. With a properly configured system (bare Linux, no GUI, some kernel sysctl tuning, etc), you should see fantastic performance on that box. Plus, with turbo you can always hit that little button[1] on the front of your server if you need a bit of a performance boost. :) --Tim [1] http://en.wikipedia.org/wiki/Turbo_button -- _ -- 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] SS7 + T1
- Original Message - I spoke with the Asterisk Pre-sales team and they said that SS7 support isn't technically supported, but it is there (e.g. talk to the OS community about this) so here's my question: I'm trying to interface an Asterisk Softswitch to a Nortel DMS100. If I get a dual-span card, can I run SS7 signaling over one span, and a T1 over the other span and have Asterisk link the two (e.g. caller-ID for the call on the 1st channel comes across the SS7... and so forth?)? I'm personally not experienced with SS7. However, Sangoma[1] does provide support for SS7 with Asterisk. You may want to contact their sales/support personnel with your questions. --Tim [1] http://www.sangoma.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] FXS - Power Alarms
Greetings- On occasion, I'm seeing the following in syslog on some systems using analog cards with FXS modules: [ 1664.861183] Power alarm on module 1, resetting! These are typically cleared by restarting asterisk/dahdi, or power cycling the system. However, I'm wondering if anyone can explain what condition exactly causes this. My best guesses: -Too high current draw on port (shorted pair) -Not enough current supplying card/module (bad power supply) -Voltage introduced to FXS port (FXS port connected to FXO or POTS line for example) Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] State of Asterisk+Virtualization+Timing
Greetings- I'm about to dive into the process of virtualizing some of my Asterisk (primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, the primary issue preventing me from moving was the lack of proper timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either OpenVZ or KVM, but each seem to have independent issues that need to be addressed: OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant access to host node timing source (physical device, or dahdi_dummy in /dev/dahdi/) to the containerized Asterisk process. KVM - Higher overhead, easier installation, 'true virtualization'. Primary issue is not timing per se, but KVM scheduling. Timing source, while present from dahdi_dummy natively may still not get proper scheduling by KVM process. This could also affect general call quality (even non IAX2 trunked voice), DTMF, etc. I have to believe there are others running virtualized Asterisk installations with some degree of success on OpenVZ or KVM. Care to share your thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Keeping Voice Call Active During Data Connectivity Loss
Greetings- I'm working on a unique Asterisk installation where I've been given a requirement of keeping a voice call active, even during a data connectivity loss. So, let's assume I have remote users connecting to an Asterisk server via sometimes unreliable connectivity such as satellite, wireless, or shudder dial-up. It is certainly possibly this connectivity will go down for a period of time anywhere from a few seconds to a few minutes (or more). During this outage, if a call was already in session, is there any way to prevent the call from be hung up, and simply kept alive until media can begin flowing again? In this situation, both sides of the link would be running Asterisk, 1.4.x or 1.8.x. Is this as simple as telling both sides not to hangup at a lack of media? Are the steps the same whether using SIP or IAX (preferred IAX in this usage case, unless SIP is specifically required)? Thanks! --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk/DAHDI with Dynamic T1s
Greetings- From time to time, I find myself working with (or customers working with) dynamic T1s. They are typically standard T1s that terminate to an Adtran device which utilizes the channels for data (64kbps X 24) until a call is pushed inbound/outbound on the circuit. One data channel is automatically peeled off the circuit (removing 64kbps from total data throughput capacity), and reallocated as a voice channel. Is it possible for Asterisk/DAHDI to handle a situation such as this? If I recall, DAHDI does have some data functions to it, but I'm not sure if it can handle the circuit as data (presented to kernel for iptables routing/nat), and/or if it can automagically reallocate channels for voice usage on the fly. Thoughts? --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.3.3 T.38 Gateway
- Original Message - On 11-09-01 07:04 AM, Tim King wrote: I have found numerous claims that 1.8 can do T.38 gateway with a patch, however I am yet to find the patch or any instructions on implementing it. Anyone have a link? Asterisk-10.0.0-beta1 is another option. I've been testing the T.38 functionality in 10.0.0-beta1 with very successful results. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycoms rebooting themselves
- Original Message - Well, we've taken the time to check out the wiring. It's only 3 years old and looks like the people who did it knew what they were doing. Nice work. Rebooting the cable modem, router, and switch didn't fix the problem. Also, we had an instance today where ALL of the phones went down within minutes of each other. The Internet connection was still active. Looks like more often than not, all of the phones die at the same time. Any other ideas? If they're all powered via PoE on the same switch, look to diagnosing the switch itself. Look for issues with heat (not enough cooling or circulation), or depending on the switch, you could be pulling too much power from the PoE module contained within. Does your switch's PoE module put out enough power for 'X' number of phones at 'Y' number of watts each? Either of these problems would lead to the switch shutting down or resetting the PoE module which causes your phone reboots. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
