Re: [asterisk-users] Fwd: add a new queue strategy: SBR
On Sun, Mar 8, 2009 at 9:44 PM, Mark Michelson mmichel...@digium.com wrote: Caution: One shortcoming of queue member penalties is that they are not taken into account if a queue member of a low penalty does not answer a call. Say for instance that the queue application determines that there are two members available to answer an incoming call. One member has penalty 1 and the other has penalty 2. If the member with penalty 1 does not answer the call, the queue application still considers that member to be available the next time that it tries to reach a member. The member with penalty 2 will only be tried if the queue application can determine *before the call is placed* that the member with penalty 1 is unavailable. For traditional queues, you can set: autopause = yes on a per-queue basis in queues.conf. If the member with penalty 1 does not answer a call, they will automatically be paused and will not receive additional queue calls until they have been unpaused (either by themselves or by an administrator). If you are using dynamic queues with Local channels (as described in doc/queues-with-callback-members.txt in the Asterisk source), you can also optionally implement this functionality directly in the dialplan. This has the added benefit of allowing you to choose on a per-agent basis who is eligible for autopause. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install spandsp from source in lenny ?
On Mon, Mar 9, 2009 at 11:13 AM, Olivier oza-4...@myamail.com wrote: Anyway, whenever I'm typing make menuselect, app-fax is greyed out as in my opinion, spandsp libriaries have not been found. Maybe, I should have typed something like (as suggested http://www.voip-info.org/wiki/view/Asterisk%20T.38) ./configure --prefix=/usr but I prefer to ask here, as I'm afraid to spoil everything giving wrong input. You can do this, which will put the spandsp library in /usr/lib where configure for Asterisk can find it, or you can run Asterisk's configure as configure --with-spandsp=/usr/local to tell it to look where you have already installed it. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: add a new queue strategy: SBR
On Mon, Mar 9, 2009 at 5:44 PM, nik600 nik...@gmail.com wrote: Thanks, i've tested and it works (1.4.23.1). Just 2 questions: 1) this approach seems to be an hack and not the implementation of a feature is it really used in corporate solutions? 2) using queue show 001 i can't see the ringing status, is that correct (In Use, Not in Use,Paused works now properly)? I've never really noticed the lack of a ringing status. Our queue setup has just worked, so I usually only have to use queue show when there's a problem. I do know that the AMI reports the ringing status. The Local/n solution has the added problem of not handling attended transfers correctly. When using a Local channel with the /n flag, if an agent performs an attended or SIP transfer, or does a 3-way call on their own phone and then hangs up, Queue() will still consider the agent In Use until the original transferred call is hung up. Maybe polling the device state using the SIP channel would be better, but as you told me this feature is available only on 1.6.x. It was backported to 1.4.19, but the patch no longer applies cleanly to newer versions. There were some locking changes just after that version. If you want to give it a try, I found it at: http://ftp.iq-labs.net/state_interface-1.4/ Then there's this: http://reviewboard.digium.com/r/116/ The corresponding func_devstate has also been backported, but it's pretty old: http://svncommunity.digium.com/view/russell/asterisk-1.4/func_devstate-1.4/ I got the 1.4.19 backport to compile against a 1.4.20.1 codebase, but Asterisk would core as soon as app_queue.so loaded, so clearly I didn't quite get it right. I eventually punted and changed my dynamic queues to just use the actual SIP/x channel names. It's been working fine for over a year now. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call file concurrency
On Fri, Feb 27, 2009 at 4:14 AM, Christian Victor christ...@victormedia.de wrote: 2009/2/27 Bill Michaelson b...@cosi.com Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? Afaik only by limiting the number of call files in the directory. If you can get the outgoing directory (or a reaonable parent) on its own mountable partition or volume, you could accomplish this with disk quotas. It won't control how many Asterisk processes at once (does it even handle them in parallel?), but it will control how many can possibly be queued up waiting to be processed. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP 650 Sidecar
On Mon, Oct 13, 2008 at 10:19 AM, Jeremy Mann [EMAIL PROTECTED] wrote: Is the IP 650 sidecar compatible with asterisk? Yes. Our attendant phone is a 650 with three expansion modules (sidecars). Asterisk can't really tell the difference. The sidecar just gives the phones more buttons for lines or speed dials. If I pair it with the IP 650 phone, can I have more than 6 lines registered w/ the server? The 650 supports up to 12 lines with an expansion module. http://www.polycom.com/usa/en/products/voice/desktop_solutions/soundpoint/desk_phones/soundpoint_ip650.html -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: headsets
On Sun, Oct 5, 2008 at 3:30 PM, Bill Michaelson [EMAIL PROTECTED] wrote: The IP330 has a subminiature jack for headset/mic combos. Are there quality headsets anyone would recommend for in-office use for heavy users with these phones? Using any wiring path? I've tried a cell phone earphone/mic, and it sounds OK, but it's flimsy for this application. We've started switching out our Plantronics M-series amplifiers and headsets with headsets from Jabra. Some of our agents have a problem with their M-series amps where they get no audio when they pick up using the phone's headset button, and they have to quickly go on-hook and off-hook again with the amp. We also had some S11 sets, with which we experienced horrible echo. The main upshot for us is that the Jabra headsets don't require an external amp, so they're simpler to install and cost less. We're using them on Polycom 330, 550, and 650 sets, and the audio quality is great. The have the usual quick-disconnect, so they're appropriate for a call center (though the connector is not compatible with Plantronics). They have adapters for both 2.5mm and RJ-8 modular plugs, so they can be used on any Polycom IP phone with a dedicated headset jack. Feel free to contact me off-list if you'd like the part numbers we're using. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
I have not applied the 1.4 backport to my system, so I haven't used DEVSTATE, but this page appears to show how to do what you want: http://www.voip-info.org/wiki/view/Asterisk+func+Devstate That page also has a link to the backport. -James On Thu, Sep 4, 2008 at 11:34 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: James, very useful info especially about enable/disable the light next to the speed dial button which is exactly what I am after. I am currently using 1.4.x and would be interested to know how this can be achieved. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfers on AgentLogin()
Since AgentLogin() essentially keeps a channel to the agent open all the time, a normal SIP transfer will do exactly as you say. That is, it will try to send the agent's login session into queue, which isn't what you want. As Matt suggested, you need to pass the t option to the Queue() application. This will let your agents perform a DTMF transfer using the codes defined in features.conf. The agent basically dials a short code while talking to the caller. Asterisk intercepts it, and then prompts the agent for the extension to transfer the call to. Look in features.conf for more information. Fair warning, I have never needed to use this feature, so I can't attest to exactly how it behaves. We use dynamic agent logins, so we've never had to deal with AgentLogin(). This allows us to do normal SIP transfers. Also, you will probably have to do one of two things in your sip.conf. One, set canreinvite to no to keep Asterisk in the call path, that way it can intercept the DTMF tones. Or, two, set dtmfmode to info, so that DTMF tones are converted to SIP INFO messages, which Asterisk will see. At least, that's how I think it works. :) -James On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton [EMAIL PROTECTED] wrote: I've tried the regular, xfer button on xlite, dial 100 (to transfer to the queue), and hit go back to line 1 and hit xfer again. But it's AgentLogin(), so it transfers the full persistent connection to the queue instead of the call itself and this causes the transferring agent to logout. Either that, or I'm doing something wrong. There is no documentation out there so I don't know how it would work for AgentLogin(). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: August 30, 2008 6:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Transfers on AgentLogin() What did you try and how did it fail? Are you using the t option in queue? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
I'm using a similar feature on 550 and 650 phones, also running 2.2.2. I've never used the attendant option to do it, though, so I'm not sure how it differs from what I'm doing. Instead, on the phones that are allowed to do this, I have the following in their XML config. You could just as easily enable it in phone1.cfg for all phones: feature feature.1.name=presence feature.1.enabled=1/ Reboot then phone, and then when you add a new entry to your speed dial directory (or edit an existing one), you will see a new Watch Buddy option, which corresponds to the bw line in the MAC-directory.xml file. The speed dial icon changes from the multiple-dots icon to a silhouette of a person or will blink when the phone is not registered, and the LED will go red when they're on a call. It still functions as a speed dial, too. John Lee was also correct that Polycom needs Asterisk's help. In extensions.conf (or .ael), you need to set a hint for any extension you want your 501 to see. In sip.conf, you need to set allowsubscribe to yes, and set subscribecontext to a context that can see those extensions. I'm using this on our attendant phone, which is a 650 with three expansion modules. The phone is programmed with several dozen employee extensions, with Buddy Watch enabled for all. This lets the receptionist see who is on the phone, so callers she transfers aren't surprised when they go to voicemail. It's not perfect, because it doesn't display DND or queue login/pause status, but it's better than nothing. -James On Thu, Sep 4, 2008 at 7:09 PM, Robert McNaught [EMAIL PROTECTED] wrote: I am using a polycom IP601, sip 2.2.2.0084 In the phone1.cfg file I set: attendant attendant.uri=4158149992 attendant.reg=1/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
I believe DEVSTATE() in 1.6 (backported to 1.4 in various places) will let you arbitrarily control BLF, so you could control it in the dialplan when an agent logs in or out (or pauses, or whatever). Separately, you might be able to use sipsak (http://sipsak.org/) to construct a SIP message that essentially forges an event to cause a BLF state change on the phone. This guy is using it to control the MWI light, so maybe it could be modified to control BLF: http://www.siliconvp.us/modules.php?name=Newsfile=articlesid=7 -James On Thu, Sep 4, 2008 at 9:53 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: It's not perfect, because it doesn't display DND or queue login/pause status, but it's better than nothing. James, on a different note, is it true that at this stage, we can never get any queue login status/light on Polycom phone? I posted a query a few days ago but I have got 0 reply. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
Sorry, needed to add one more note. To clarify, my agent phones have a speed dial assigned for their login, and another to pause/unpause. I could then use DEVSTATE to enable or disable the light next to their speed dial button based on their status. I can't use it to update anything on the LCD screen. Today I do it completely differently. I use the idle window minibrowser, and each agent phone has its own page it loads. I wrote a perl script that connects to the AMI to watch the status of our agents, and for any status change, it updates this page to reflect their status. Since Polycom doesn't let you push data out to the phones, they have to poll on a regular interval. I think ours are set to every 5 seconds. It's a hack and it's ugly, but it works. -James On Thu, Sep 4, 2008 at 10:38 PM, James Sneeringer [EMAIL PROTECTED] wrote: I believe DEVSTATE() in 1.6 (backported to 1.4 in various places) will let you arbitrarily control BLF, so you could control it in the dialplan when an agent logs in or out (or pauses, or whatever). Separately, you might be able to use sipsak (http://sipsak.org/) to construct a SIP message that essentially forges an event to cause a BLF state change on the phone. This guy is using it to control the MWI light, so maybe it could be modified to control BLF: http://www.siliconvp.us/modules.php?name=Newsfile=articlesid=7 -James On Thu, Sep 4, 2008 at 9:53 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: It's not perfect, because it doesn't display DND or queue login/pause status, but it's better than nothing. James, on a different note, is it true that at this stage, we can never get any queue login status/light on Polycom phone? I posted a query a few days ago but I have got 0 reply. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Sun, Aug 31, 2008 at 1:45 AM, C F [EMAIL PROTECTED] wrote: No, in the beginning you asked because you don't have the experience so folks like myself that do have the experience answered. It might work for you, no one knows and you THINK it will work, it's a hit and miss, stability is huge issue, thats where experience comes in. If you want something that I or the other people here just think works, then just get an ATA. If you want something we have experienced and know that it works, then get a channel bank. Your points are well taken, and I apologize for coming off as presumptive on the subject. It wasn't intended that way. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Faxing through Zap cards
I think I have this straight, but I wanted to bounce it off anyone who might be more knowledgeable. We are installing an Asterisk server at a location that only has PRI. Inbound fax comes in on the PRI with its own DID. Currently, the PBX handling it just has a PRI port and sends calls for that DID to an FXS port that the fax machine is connected to. My plan was to use a Digium TE card for the PRI and a TDM card with an FXS port to connect the fax to. I know I'll need to use the fax detection feature to disable echo cancellation. I can't use the Zaptel dacs features because I won't know ahead of time which channel the fax calls will come in on (though I could potentially use it for outbound faxes). So the call path would look like this: PSTN --- PRI --- TExxx --- Asterisk --- TDMxxx FXS port --- fax machine Does this sound reasonable? Any gotchas I should watch out for? -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 10:31 AM, Doug Lytle [EMAIL PROTECTED] wrote: Yes. You may also want to look into HylaFAX+, it's a wonderful piece of software that will allow you to have more control over your faxes. Thanks for the input, Doug. Moving away from standalone fax machines to a fax server, such as HylaFAX, would certainly be nice as well. It's more of a logistical problem for us, in that some of our offices just don't have document scanners, and they frequently have to fax paper documents. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 10:47 AM, Gordon Henderson [EMAIL PROTECTED] wrote: Just make sure you get the IRQs separated and you might want to think about a TDM410 rather than a TDM400 card. Thanks, I'll keep that in mind. We're buying new hardware for this, so we can spec it out accordingly. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 11:06 AM, C F [EMAIL PROTECTED] wrote: I would suggest that you go with a dual port T1 card and a channel bank. you will have much greater fleibality as well as more stable connection. Thanks for the tip. If faxing in this setup does prove to be unreliable, this is a good backup. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Fri, Aug 29, 2008 at 4:02 PM, C F [EMAIL PROTECTED] wrote: No it's not a backup, it's the preferred way for lots of reasons. The ones that come to mind: 1. Same PCI slot will be able handle both, which means if properly configured the RTP will not travel the MB hence no IRQ to worry about. 2. One Adit 600 gives you for around $400.00 on eBay 24 FXS ports. 3. You can have POTS on that Adit 600 as well, which will allow you to use some lines for backup etc. I get what you're saying, but... 1. We already know the slots that the cards will go in have their own IRQs. 2. Twenty-four FXS ports is overkill for two faxes. :) 3. Believe it or not, but the office this is going into has no POTS lines, just PRI and T1. So in the end, I think we have a solution that will work, and if it doesn't, I will take your (and CF's) advice with the channel bank. Thanks. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy not working - transmit frame type 64 warning
On Thu, May 15, 2008 at 10:47 AM, James Sneeringer [EMAIL PROTECTED] wrote: When I try to use ChanSpy, the following message is sent repeatedly to the console (wrapped for readability): WARNING[32125]: chan_sip.c:3709 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) I see I have you all stumped! Let's try this from another angle. Is anyone successfully using ChanSpy on 1.4.x? I've tested on most of the recent releases, including yesterday's 1.4.20.1 release. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root
On Sat, May 17, 2008 at 9:21 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, May 16, 2008 at 06:32:30PM -0500, James Sneeringer wrote: The safe_asterisk script monitors the actual asterisk process, and if it dies for some reason, Not for some reason. For instyance, if asterisk decides to die the script should not restart it. And if it got a SIGTERM? (e.g.: from init on shutdown?) True, I oversimplified a bit. I should have said it restarts Asterisk if it dies abnormally (e.g. with a return code of anything other than zero). Init shouldn't have to kill it if the startup script can also shut it down cleanly (which in my case on Ubuntu means stopping safe_asterisk first). Also note that asterisk.conf options override command-line options (and not the other way around, as you might have learned to expect from most other applications). Some asterisk.conf options, such as runuser and rungroup, don't appear to work at all. I can get Asterisk to run non-root using -U and -G on the command line, but attempting to do it in asterisk.conf instead doesn't work for me. The command line is good enough for me, so I haven't taken the time to figure out why it doesn't work. Question: what does it take to move the voicemail file from /etc/asterisk/voicemail.conf to /etc/asterisk/writble/voicemail.conf ? Patch voicemail.conf and leave a compatibility symlink for the others? Yes, you would have to patch those two applications. The filename itself is hardcoded into app_voicemail.c and app_directory.c. It picks up the path from astetcdir in asterisk.conf, or uses /etc/asterisk if that option is not defined. A simple symlink isn't good enough because when Asterisk rewrites the config file (after a user changes their passcode), it unlinks the old one without checking whether it's a symlink it needs to dereference. I haven't looked too deeply into the code, so I don't know if it's as simple as changing the VOICEMAIL_CONFIG to subdir/voicemail.conf. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root
On Fri, May 16, 2008 at 3:04 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: First of all, thanks Philipp, Alan, Tzafrir and James for your valuable comments. I have listed below the exact list of commands to run for reinstalling asterisk 1.4.* as non-root on a Redhat / Fedora distro. Hope others can benefit. I have the following comments/questions though: 1) #What is safe_asterisk used for actually? I did not touch it in my modification because I don't know when is it triggered? The safe_asterisk script monitors the actual asterisk process, and if it dies for some reason, it restarts it and optionally notifies you. It's just a precaution. MySQL is often run under a script called mysqld_safe for the same reason. 2) #I do not actually know whether we really need to modify /etc/asterisk/asterisk.conf? Is this file read by asterisk at all? Seems like an important file name - asterisk.conf? It is read by asterisk, but whether you need to change any of the defaults really depends on your environment. Most of the options in it have equivalent command-line options, so you might want to use asterisk.conf instead of modifying the startup script (which could be overwritten the next time you upgrade). 4) There is an additional chmod to run for letting voicemail.conf to be written by group asterisk. What I found was that /etc/asterisk also needs to be writable by the asterisk user, because asterisk will unlink and recreate the file, so it needs to be able to write to the directory, not just the file. You can protect yourself a little bit by setting the sticky bit on /etc/asterisk, so even if asterisk goes nuts, it can't whack files it doesn't actually have write permissions on. chmod g+w /etc/asterisk/voicemail.conf chmod g+w,+t /etc/asterisk -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy not working - transmit frame type 64 warning
When I try to use ChanSpy, the following message is sent repeatedly to the console (wrapped for readability): WARNING[32125]: chan_sip.c:3709 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4) This appears to happen because SLINEAR (frame type 64 / 0x40) is not considered a native format for the channel I'm trying to spy on, so sip_write() bails. DEBUG[32006] chan_sip.c: *** Our native formats are 0x4 (ulaw) DEBUG[32006] chan_sip.c: *** Joint capabilities are 0x4 (ulaw) DEBUG[32006] chan_sip.c: *** Our capabilities are 0x46 (gsm|ulaw|slin) DEBUG[32006] chan_sip.c: *** AST_CODEC_CHOOSE formats are0x4 (ulaw) The symptom is that the spying listener hears nothing on his or her handset. The spied-on channel can continue to communicate with the other end without issue. If I tell ChanSpy to record the call with the r() option, a .raw file *is* created and does contain the complete audio stream, so it's at least partly working. The problem appears to be in converting back from SLINEAR to the spying listener's native format. I have format_sln.so loaded, and I have it listed as a valid codec for all of the SIP devices involved (Polycom 330's and a 550 doing the spying). Has anyone else run across this? Maybe there's something really simple I'm missing? -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Asterisk: Install Asterisk as non-root
On Thu, May 15, 2008 at 5:30 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, May 15, 2008 at 06:17:12PM +1000, Lee, John (Sydney) wrote:
5) Another article says that running as non-root will prevent ToS being
used. What is ToS? Do I need to be concerned?
Anybody wants to write something about this?
I recall a change in that area in recent Asterisk 1.4-s .
ToS is supported when running non-root on Linux by using kernel
capabilities. On Ubuntu, the libcap-dev package is required for this.
It provides libcap.{a,so} and sys/capability.h, which the Asterisk
configure script will check for before you compile. You can check to
see whether your binary is linked against libcap using the ldd
command:
$ ldd /usr/sbin/asterisk
linux-gate.so.1 = (0xe000)
libdl.so.2 = /lib/tls/i686/cmov/libdl.so.2 (0xb7fd9000)
libcap.so.1 = /lib/libcap.so.1 (0xb7fd5000)
libpthread.so.0 = /lib/tls/i686/cmov/libpthread.so.0 (0xb7fc2000)
libncurses.so.5 = /lib/libncurses.so.5 (0xb7f81000)
libm.so.6 = /lib/tls/i686/cmov/libm.so.6 (0xb7f5f000)
libresolv.so.2 = /lib/tls/i686/cmov/libresolv.so.2 (0xb7f4c000)
libc.so.6 = /lib/tls/i686/cmov/libc.so.6 (0xb7e1d000)
/lib/ld-linux.so.2 (0xb7fe5000)
-James
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Re: [asterisk-users] Polycom IP 330 w/VLAN?
The 330/550/650 phones have a built-in 2-port switch that speaks 802.1q. Usual use of this is to send two VLANs down the wire. The phone is configured to use one, and the phone transparently passes the other to the phone's PC port. On Cisco, this would be a trunk port with two VLANs, one for the phone and one for the PC (the latter usually being the default VLAN). On HP ProCurve, you assign one tagged and one untagged VLAN to the phone. The VLAN used by the phone can be configured in several ways: 1. Hard-code it on the phone. Not recommended if you have lots of phones. 2. Auto-discovery using CDP. Requires Cisco or older HP switches. 3. Auto-discovery using DHCP. Disabled by default in SIP 2.1.x. We use the third option. The phone first does DHCP on the default VLAN (since it does not yet know which VLAN to use). The DHCP server sends back the VLAN to use, which cause the phone to send a DHCP release, reconfigure itself for the new VLAN, and do DHCP again on the new VLAN. We use this method because we have newer HP switches that have removed CDP support in favor of LLDP, which Polycom does not (yet?) support. -James On 3/11/08, Ken D'Ambrosio [EMAIL PROTECTED] wrote: Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP600 + PC share same switch port withVLAN
Glad it worked for you. The warning is normal for Cisco switches. Basically, the portfast setting disables Spanning Tree (802.1d friends) negotiation for the ports. It is only dangerous if you actually connect that port to another bridging device that could potentially have an alternate layer-2 path back to the switch (i.e. a loop). In the case of your Polycom phones, it is completely safe. It would only be unsafe if you connected both ports of your phone (the uplink *and* PC ports) into the switch, which you are unlikely to do. It's up to you, of course. Without portfast, you're looking at about 30 seconds for STP to negotiate whenever the port bounces, during which time higher layer protocols are unavailable. This may interfere with CDP and DHCP, if you're using those. -James On Mon, Mar 3, 2008 at 8:57 PM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: James, thanks for the suggestion. I am just using native vlan and I did what you said and I believe it works :-) interface FastEthernet2/0/2 description VOIP VLAN 100 switchport trunk encapsulation dot1q switchport trunk allowed vlan 1,100 switchport mode trunk duplex full speed 100 However, when I entered spanning-tree portfast trunk, I received a warning message which says: Warning: portfast should only be enabled on ports connected to a single host. Connecting hubs, concentrators, switches, bridges, etc... to this interface when portfast is enabled, can cause temporary bridging loops. Use with CAUTION As the polycom phone is acting like a switch, I decided not to put that option in. ***Also, can I just confirm that with the current QOS (quality of service) settings on the polycom phones, the phone should have priority over the PC? Thanks in advance. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Sneeringer Sent: Saturday, 1 March 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port withVLAN As far as I can tell, with Polycom phones you cannot do what you're asking (which is for the PC and the phone to be in the same VLAN while the PC is connected to the phone). I don't know how they handle it when the voice frames are untagged, but they definitely won't pass tagged voice frames to the PC port: http://knowledgebase.polycom.com/KanisaPlatform/Publishing/616/12526_f.S AL _PUBLIC_1_2.html Your switch port is configured for untagged frames on a single VLAN (that's what access mode is). Polycom phones need voice and data to be on separate VLANs in order for you to use the PC port. Since Polycom phones apparently don't support the Cisco Voice VLAN feature, you need to configure the port as a trunk port, which will allow you to send multiple VLANs to the phone. The phone will take frames tagged for your designated voice VLAN, and will pass the rest on to the PC port. For example: interface FastEthernet2/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan XXX switchport trunk allowed vlan XXX,100 switchport mode trunk spanning-tree portfast trunk Replace XXX with whatever your PC VLAN is. Setting XXX as the native VLAN for this port will cause frames in that VLAN to be untagged for that port, which is what your PC probably expects. If it happens to be 1, then it's the native VLAN by default. The last command may or may not be available, depending on your version of IOS. If it isn't, portfast just won't work and you're just stuck with STP negotiation anytime the port bounces. -James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN
As far as I can tell, with Polycom phones you cannot do what you're asking (which is for the PC and the phone to be in the same VLAN while the PC is connected to the phone). I don't know how they handle it when the voice frames are untagged, but they definitely won't pass tagged voice frames to the PC port: http://knowledgebase.polycom.com/KanisaPlatform/Publishing/616/12526_f.SAL_PUBLIC_1_2.html Your switch port is configured for untagged frames on a single VLAN (that's what access mode is). Polycom phones need voice and data to be on separate VLANs in order for you to use the PC port. Since Polycom phones apparently don't support the Cisco Voice VLAN feature, you need to configure the port as a trunk port, which will allow you to send multiple VLANs to the phone. The phone will take frames tagged for your designated voice VLAN, and will pass the rest on to the PC port. For example: interface FastEthernet2/0/1 switchport trunk encapsulation dot1q switchport trunk native vlan XXX switchport trunk allowed vlan XXX,100 switchport mode trunk spanning-tree portfast trunk Replace XXX with whatever your PC VLAN is. Setting XXX as the native VLAN for this port will cause frames in that VLAN to be untagged for that port, which is what your PC probably expects. If it happens to be 1, then it's the native VLAN by default. The last command may or may not be available, depending on your version of IOS. If it isn't, portfast just won't work and you're just stuck with STP negotiation anytime the port bounces. -James On Fri, Feb 29, 2008 at 12:13 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Thanks very much for the quick response. However, switchport voice vlan.. I thought is only valid for CISCO phones and I am using Polycom and thus it would not work. Furthermore, I have already tried switchport voice vlan... before I emailed to the list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob G Sent: Friday, 29 February 2008 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN You can paste and copy innterface FastEthernet2/0/1 switchport access vlan 20 switchport mode access switchport voice vlan 120 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree portfast - Original Message - From: Lee, John (Sydney) To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN Date: Fri, 29 Feb 2008 16:39:24 +1100 Hi all, I have been googling and testing without any luck and would appreciate any guidance from anyone. A port has already been configured on the CISCO switch with the following: interface FastEthernet2/0/1 description VOIP VLAN 100 switchport access vlan 100 switchport mode access duplex full speed 100 I plugged the phone into the port and everything worked as far as VOIP is concerned. Then I plug a PC into the PC port of the Polycom phone with the hope that I only need one port to support 2 devices. (I wanted the VOIP phone to use VLAN 100 and PC just the native VLAN) PROBLEM: However, I found that I could not get the PC (using DHCP) to get an IP address at all. It seems to be that the traffic from the PC is also tagged as VLAN 100 as well. I was told by others that there is a setting on the Polycom phone which allows the traffic of the PC, under this type of settings, to go native. Can anyone please help? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Want an e-mail address like mine? Get a free e-mail account today at www.mail.com! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
