[asterisk-users] Working Config for Polycom VVX and Auto Answer
Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX phones work.
I tried the guide here:
http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167
And I have this in my diaplan:
exten = _8XX,1,SIPAddHeader(Alert-Info: info=ringAutoAnswer)
exten = _8XX,2,Dial(SIP/${EXTEN:1},20,tk)
But whenever I attempt a call to a matching exten, it just rings
normally for the 20 seconds I have indicated here and never answers.
I found a setting in the phone's GUI: Auto Answer SIP Calls. When I
set this to Yes. It will auto answer, but it auto answers ALL
calls, not just ones with the Alert-Info header set.
Any guidance is appreciated.
Thanks!
Noah
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Re: [asterisk-users] Working Config for Polycom VVX and Auto Answer
On Fri, Mar 14, 2014 at 12:36 PM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi -
Just wondering if anyone has gotten a Polycom VVX phone to
successfully do an Auto Answer with asterisk. I have an older
generation of Polycom phones that do this just fine, but I can't seem
to make the VVX phones work.
I tried the guide here:
http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167
And I have this in my diaplan:
exten = _8XX,1,SIPAddHeader(Alert-Info: info=ringAutoAnswer)
exten = _8XX,2,Dial(SIP/${EXTEN:1},20,tk)
But whenever I attempt a call to a matching exten, it just rings
normally for the 20 seconds I have indicated here and never answers.
I found a setting in the phone's GUI: Auto Answer SIP Calls. When I
set this to Yes. It will auto answer, but it auto answers ALL
calls, not just ones with the Alert-Info header set.
Any guidance is appreciated.
Thanks!
Noah
Well, in case anyone else is interested, it's working now. I must
have mistyped something the first time around because it is now
working with the exact settings I describe above.
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[asterisk-users] S110M not working
Hi All - I pulled from a working system a TDM400 with one s110 fxs and three x100 fxos. I put it into a new box and the fxs no longer works. The fxos work just fine. I thought it was odd, but I chalked it up to a random chance failure and ordered another s110. The replacement doesn't work either, and now I'm confused. dahdi_scan recognizes the s110, but it says it fails. Output looks like this: [2] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 05 Slot 01 basechan=5 totchans=4 irq=169 type=analog port=5,FXO port=6,FXO port=7,FXO port=8,FXS FAILED I tried moving both the s110s to other positions on the TDM400, but it fails in all of them. Is this a power issue? The system it's in only has sata power connectors, so I had to splice on an ide power connector. Both the 5v and 12v lines are connected through all the way from the power supply to the tdm400 card. If I force the issue, and manually configure the s110 like this: --- fxoks=8 echocanceller=mg2,8 --- the dahdi startup script throws this error: Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 8: Invalid argument (22) Selected signaling not supported Possible causes: FXO signaling is being used on a FXO interface (use a FXS signaling variant) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span Have I bungled the power connection and/or the config, or am I just extremely unlucky and have two bad s110s? Thanks, Noah -- _ -- 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] S110M not working
Hi. FXS cards use FXO signalling, and vice versa. Think of it this way: FXS cards want to look like a CO when talking to stations, and FXO cards want to look like a phone when talking to a CO. Thanks, Barry. I am aware of this. You'll notice in the config line that I used fxo signalling. I'm just thinking that the failure that dahdi_scan see may be because the s110 isn't getting power. On Tue, Nov 16, 2010 at 1:46 PM, Barry Miller asterisk-us...@notanet.net wrote: On Tue, Nov 16, 2010 at 01:17:08PM -0500, Noah Miller wrote: Hi All - I pulled from a working system a TDM400 with one s110 fxs and three x100 fxos. I put it into a new box and the fxs no longer works. The fxos work just fine. I thought it was odd, but I chalked it up to a random chance failure and ordered another s110. The replacement doesn't work either, and now I'm confused. dahdi_scan recognizes the s110, but it says it fails. Output looks like this: [2] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 05 Slot 01 basechan=5 totchans=4 irq=169 type=analog port=5,FXO port=6,FXO port=7,FXO port=8,FXS FAILED I tried moving both the s110s to other positions on the TDM400, but it fails in all of them. Is this a power issue? The system it's in only has sata power connectors, so I had to splice on an ide power connector. Both the 5v and 12v lines are connected through all the way from the power supply to the tdm400 card. If I force the issue, and manually configure the s110 like this: --- fxoks=8 echocanceller=mg2,8 --- the dahdi startup script throws this error: Running dahdi_cfg: DAHDI_CHANCONFIG failed on channel 8: Invalid argument (22) Selected signaling not supported Possible causes: FXO signaling is being used on a FXO interface (use a FXS signaling variant) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span Have I bungled the power connection and/or the config, or am I just extremely unlucky and have two bad s110s? Hi. FXS cards use FXO signalling, and vice versa. Think of it this way: FXS cards want to look like a CO when talking to stations, and FXO cards want to look like a phone when talking to a CO. -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] S110M not working
I'm just thinking that the failure that dahdi_scan see may be because the s110 isn't getting power. If you see FAILED in dahdi_scan for the FXS port, then most likely there will be some indication of what actually failed in the kernel log. Is there anything in dmesg? Aha! Thanks, Shaun. dmesg says: ProSLIC on module 3 failed to powerup within 501 ms (0 mV only) That certainly looks like a power issue to me. So, next question, why is it not getting power if all the leads are connected? I guess it's time to do the get out the current meter. Thanks! Noah -- _ -- 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 Query
Hi Garge -
exten =
,1,Asterisk_Application(Action) ;Dial(Zap/1/${Phone_Number_you want})
Two things:
1. There is no such thing as Zap anymore. Zap has been renamed to
Dahdi because of a trademark issue. So your extension should look
like:
exten = ,Dial(Dahdi/1/)
2. Do you really mean to dial ''? This number should be a valid
phone number.
- Noah
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Re: [asterisk-users] OT: NAT in SPA922
Ok..So what ip phone model do NAT? I think you'd struggle to find one. If it's a requirement you're probably doing something wrong... Definitely get a router. Plug the IP phone into the router, and then you can plug the computer into the phone or the router. - Noah -- _ -- 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: NAT in SPA922
It is a building, with 24 separated rooms, each room will have a PC and a IP Phone. Every room connected to a switch Cisco 2950. I want keeping all PCs isolated behind a NAT (no access to neighbour's PC), and still keep communication in same LAN between all IP Phones. Should I take another approach on that? Put each PC in its own VLAN. Keep all the phones in one VLAN. Although having a $30 router in each room hanging off the phone would accomplish what you want also. Take j's suggestion to use VLANs. This is not a good situation for NAT. Cisco 2950's can do VLANs. - Noah -- _ -- 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] Transfer calls using ##
I have a question about the blind transfer using ##. This works great on our cordless phone, but there have been occasions that we can't transfer using ##. I was able to reproduce the issue by doing the following: 1) Call in from the outside line, 2) Ask the operator to transfer me to an extension using ##. 3) Get the voice mail greeting of the individual. 4) Hit 0 for the operator before the greeting completed. 5) Ask the operator to transfer me again using ##. 6) Operator can't transfer and I can hear the pressing of the keys. Why can't I transfer the call the second time around? How can I fix this? The dial statement in your 'o' extension must have the 't' flag. - Noah -- _ -- 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] Echo issue
I think you need to remove the line echocanceller in system.conf You could also try to use fxotune, it'a really improving things. You also need to put echocancel=yes in chan_dahdi.conf This is a PRI, so fxotune is not the thing to use in this case. - Noah ___ -- 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] max. no. of conferences supported
What are the limits with asterisk server running on one decent (4GB, 4 CPU etc.) machine. There are a LOT of factors involved. You will likely have to do your own testing with just the specific features you want. How many MeetMe conferences it can support? What is the limit of number of participants per conference? Are you doing any transcoding? What technologies are the participants using (dahdi, sip, iax, etc)? If you're doing a conference with only sip participants and no transcoding, on the hardware you mention you should be able to comfortably host a conference with 100 participants, possibly more. I can't help you out with specific numbers, as none of the systems I administer do conferences larger than this. As for the number of conferences, I've seen one system with similar hardware specs regularly host a dozen conferences without issue. Most of these conferences have between 5 and 10 participants. Is it possible to support 1000 users in Asterisk? What is the kind of hardware needed for this? Yes, there are a good number of asterisk installations with more than 1000 users. In a recent interview with Mark Spencer, he mentioned an installation with 150,000 users. What kind of hardware all depends on what features/services you want to provide. All I can say is that you should pick and choose features/services carefully if you intend to have a lot of users. By default, asterisk will enable everything. Change it to only enable what you need. - Noah ___ -- 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] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? What kind of settings do you recommend for the txgain and rxgain? Ideally, you will need to measure to find out what settings you want. See this page on the wiki (see the note on values for PRI circuits): http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment (use dahdi_monitor instead of ztmonitor) You can also just experiment with different values. Change just one setting at a time, and then reload Dahdi. Try this to start: txgain = 0.0 rxgain = 1.0 and then on the asterisk cli, enter: module reload chan_dahdi.so If that doesn't help, try increasing to rxgain=2.0. Keep going until it sounds better. You may want to try negative values for txgain. Do I make the gain changes in chan_dahdi.conf? Yes. Make sure to put them before your channel numbers. You can specify values on a per-channel basis. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? - Noah ___ -- 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] Realtime Database Tables
I'm actually there, but I was wondering if the tables there are up to date and if any changes took place. I see all kinds of comments about changes. You could go ahead and install and then look at the table structure using your dbms. - Noah ___ -- 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] multiple sip trunks
I assume if all the SIP trunks are to the same host/port, Asterisk cannot distinguish which trunk is active when an incoming call is made- it will dump all incoming calls to the context specified in the last trunk entry of sip.conf No. SIP uses authentication (well, I guess you can not use authentication). Asterisk (and almost any SIP gateway) will correctly match the call to the trunk based on the authentication. Even if you didn't send any authentication info, asterisk will try to match the call as a guest call. It is common practice to not allow unauthenticated SIP traffic. - Noah ___ -- 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] Echo issue
Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah ___ -- 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] Free Polycom Provisioning Tool
In 2007, I released a Polycom Provisioning Tool. I retired the package earlier this year, and have had so many requests for it, I have revived the concept, new, improved, and still FREE. Any chance of you releasing the source? The asterisk GUI does Polycom phone provisioning, and that source is definitely available. - Noah ___ -- 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] Max how many users in sip.conf
I’m running CENTOS 5.3 with apache 2, asterisk 1.4.26.2, mysql 5 and php 5.2.11. top shows 928mb out of 1035mb in use with idle asterisk and 17 users. There could be a problem, but I’m relatively new to CENTOS, so any suggestions would be happy. I use CentOS for asterisk boxen, too, and my first task after installing the OS is always to use chkconfig to disable the many totally unnecessary processes that are on by default. I can usually get it down to around 400MB - 600MB used, including asterisk (mostly small offices with less than 20 users). I never use mysql or any other dbms, though. - Noah ___ -- 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 500 format file system on every reboot
Hi Warren - I have one client that is telling me that their Polycom 500's format the file system every time they reboot, and also that they are unable to make changes locally on the phone itself, only via the config files. If the config file is not available when they try to boot the phone, then they receive an error about not being able to find the config file and then the phone will not boot up. Has anyone seen anything like this before? Yes, I've seen this on a number of 500's running very recent versions of the firmware and bootrom. It only seems to affect a small number of the 500's I've worked with, though. Many of them are fine. It hasn't been a big deal for anybody as I always do provisioning through FTP, and the phones rarely need to be rebooted. I'm trying desperately to get rid of all the 500's I have out in service. Just so many bugs on them. - Noah ___ -- 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] IAX2/SIP hard phones
Hi Blaz - Do you maybe know for a fairly good quality IAX2/SIP hard phones in up to 40 USD? I don't think there are any IAX hardphone in production anymore. You might be able to find a used Atcom 320, but probably not for anywhere close to $40. It looks like voipsupply.com has some old Cisco 7910s for $40. http://www.voipsupply.com/cisco-cp-7910g That's about the lowest price you're going to find for a hardware IP phone. You should be able to get an Aastra M9116 or a Grandstream BT201 for around $50. - Noah ___ -- 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 retrieve call from hold
Hi Mike - I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and get it back. It goes on hold just fine. But when I press the resume button, nothing happends. Anyone seen this befor? Any ideas on where to start to fix it? Nope, never seen that one, and I've worked with a LOT of Polycoms. Which SIP/bootrom versions? What asterisk version? Maybe the resume soft button is programmed to do something else other than take the call off hold? What happens when you press the physical hold button (to take the call off hold)? - Noah ___ -- 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] Questions about static
We have swapped out the phone multiple times for the user. Only one user. Bad PoE port on the switch? How about local interference that the user cannot control? Does the same phone experience static when moved elsewhere? Do you have a power brick for the phone so you can try it as non-PoE? Is the static consistent or intermittent? - Noah ___ -- 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] Questions about static
We swapped PoE switches, phones, cable and switch ports multiple times. What do you mean by local interference? Cell phone? The person swears nothing is near the phone. There are lots of things that can cause interference. Radios, elevators, bad electrical wiring, you name it. Is the static still there when you move the identical phone elsewhere? If not, then the static is most probably caused by some local interference where the user is. - Noah ___ -- 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] Restricting transfers between SIP phones
So, does anyone know of a way to detect whether a call from a SIP phone is the first step of an attended transfer or an original call? It could probably work if you put a SIP proxy in between (ref. Kamilio). Another way might be to set up a special transfer extension that all users use to perform transfers. To do a transfer, all users would first transfer to that special transfer extension. The transfer extension could then read the intended destination and compare the source and destination in a series of GotoIf statements. The GotoIf statements would check the source and destination of the transfer, and if it's ok, use the transfer() app. If not, playback a message that the transfer is not allowed. It means a lot of very specific dialplan logic, and a change of procedures for the users, but it's one way to do it. - Noah ___ -- 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] hardware echo cancellation
If I get an echo cancellation module for my Digium TE121 card, will I need to do any adjustments/configuration in Asterisk? You should probably still set the gain using rxgain and txgain. IME, it's much easier setting gains on a PRI than it is on a POTS line, though. I've worked with a couple of PRIs that need no adjustments at all. Is the hardware better than the software version? The hardware version is the same algorithm as the HPEC echo canceler. It's quite a bit better than the MG2 algorithm that comes free with asterisk and maybe slightly better than OSLEC. The convergence time of the hardware algorithm is pretty fast (time it takes for the EC to effectively get rid of echo on a call). FYI: If you're considering running the software-based HPEC for all channels on a T1/E1, you should use a reasonably fast machine, as it uses quite a bit of CPU. That's one big reason to get the hardware module. - Noah TIA! ___ -- 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] Connect two Asterisk Server in IAX ?
I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. To clarify and expand on Aggio's response. You either need to have a peer and user on both machines, or you can set it up as type=friend, which is the peer and user combined. - Noah ___ -- 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 many lines do you use.
I use two ‘lines’ though ‘Line appearances’ would be a better term, though still confusing in my book. I have five line appearances on the Snom190 on my desk. I regularly use two line appearances, and on occasion, I have used three to juggle back and forth between calls. I would guess that a busy receptionist might have to use up to 6 line appearances all at once, but I can't imagine one person being able to use much more than that. I think most people get those sidecar units to do speed dials or to monitor other extensions. It's an interesting question, though. I regularly recommend Polycom 550s to my clients, but I would guess that 450s or even 335s would be just fine for most people. - Noah ___ -- 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 question
When typing 'help' on the command line (* console) is there a way to keep it from just scrolling most of the information off the top of the screen? I can't hit ctrl-s fast enough so I miss most of the info. This makes 'help' be not much help. my default scroll back buffer is set to around 1000 usually enough to capture what I need, plus you can cut paste between screens You could also make it much simpler and just set your verbosity very low or just turn it off, so there are very few messages coming across your screen. Unless you're on a really busy machine, you should be able to read most of the help screens. core set verbose 0 - Noah ___ -- 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] Linux/Asterisk on game consoles?
I don't know much about game consoles, and I was wondering if someone had successfully ported Linux and Asterisk to the current hardware, ie. Nintendo Wii, Sony PS3, or Microsoft XBox360? The Xbox is an x86 machine, so running linux and/or asterisk on it should not be too difficult. There's even a not-so-difficult method of adding a USB port, which would allow you to attach Xorcom hardware for PSTN connections. The Xbox360 is a PowerPC machine. I don't know what the status of having it run *nix is, but there's a site dedicated to it here: http://www.free60.org For the wii, there's: http://wiibrew.org/wiki/Wii_Linux - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Where to find IMAP storage doc ?
We're also working fine with it but I also do not know what the available imapflags are and what they mean. I have seen notls and novalidatecert. Out of curiosity, I spent the last 20 minutes googling for information on c-client imapflags and didn't find any definitions or even a simple list, either. There is a list of flags in the c-client man page but they seem to be a different set of flags. Let me know what you find as I would like to know what functionality and options they give us. I'd recommend compiling c-client from source. I've never run Lenny before, but I had a number of issues with various pre-compiled versions of c-client. I feel your pain on lack of documentation for compiling from source, though. The magic steps for me on CentOS were: 1. Modify the EXTRACFLAGS line of the uw-imap makefile: EXTRACFLAGS=-DDISABLE_POP_PROXY=1 -DIGNORE_LOCK_EACCES_ERRORS=1 -I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall -Wno-pointer-sign -Wno-parentheses (I think this is all I had to modify, but I can send you my complete working Makefile, if you like). 2. Compile for your platform: For lenny, I think it would be: make ldb 3. For asterisk, manually configure the location of uw-imap: ./configure --with-imap=/path/to/imap - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Testers Wanted for IMAP Voicemail patch
Hi All - At Leif's suggestion, I'm soliciting testers for a patch to IMAP voicemail. Currently, when asterisk checks for voicemails in an IMAP folder, it only looks for messages in the same context and with the same voicemail box number as the person dialing in to VoicemailMain(). I believe this artificially limits what can be done with IMAP voicemail. For example, I'd like to have an administrator who can drag and drop messages using an IMAP client from his/her voicemail account to other users' voicemail accounts. This is not possible with the current implementation of IMAP voicemail. The patch under this bug: https://issues.asterisk.org/view.php?id=15670 changes the VoicemailMain() app to look for any voicemail messages regardless of what context or user the message was originally created for. I'd love to see this make it into some version of asterisk sooner rather than later. Comments and suggestions are welcome. FYI: The patch is incredibly simple and small so stability issues should not be a concern. Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Autodial not waiting for voicemail
Hi All - I'm setting up a corporate emergency broadcast system that uses an autodialer to contact all company employees. Everything works fine except if the auto-dialed calls go to the end users' voicemail. If that happens, asterisk starts playback of the emergency message while the voicemail system on the other end is playing its outgoing message. The result is that the beginning (or all) of my emergency message is clipped off. It seems like overkill to try and use DSP to detect if the call has reached voicemail (detect the beep?), but I can't think of any other reasonable way to get the full message to the end users' voicemail. I guess I could prepend a welcome message just to kill some time while the user's outgoing greeting is playing, but that's still somewhat unreliable, especially if the user has a long outgoing message. Has anybody found a way to deal with this? Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] HPEC VPM ?
Hi - I had a client recently move their asterisk system (asterisk 1.4.26, dahdi 2.2.0.1, aex800 w/vpm module) to a new location, a building that's nearly 150 years old. I was not personally able to go there, but the person who did the move said the building's demarc room was scary-- water leaks, jumbled and frayed wiring, and all sorts of other fun. The echo on their POTS lines has proven to be quite problematic. The hybrids are balanced, txgain and rxgain are optimized individually for all channels, and the vpm module on the card is doing its job. For many calls, this has been effective. Still, echo remains on calls to some destinations, particularly those on the closer exchanges. On calls to one particular number, if I turn the echo canceler off, the echo sounds as loud and clear as if the destination was actually echo(). With the echo canceler on, echo is still very pronounced. The echo tail is clearly longer than 16 ms. I even tried disabling the vpm module (vpmsupport=0 in base.c) and using oslec instead with a setting of echocancel=512. After a long convergence period, oslec seemed to do a slightly better job than the vpm module, but echo was still bad enough to make a conversation nearly impossible. My question for anyone with knowledge on this: would HPEC do a better job than the VPM module (or oslec)? Can HPEC cope with very long echo tails? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] HPEC VPM ?
My question for anyone with knowledge on this: would HPEC do a better job than the VPM module (or oslec)? Can HPEC cope with very long echo tails? HPEC and the Digium VPMADT032 use the same algorithms from the same vendor. Aha. Thanks for this tidbit, Kevin! Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] HPEC VPM ?
Next question: does anybody know how to handle extremely long tail echo that a VPM module cannot? How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can handle 128ms echo tails, which is pretty darn long. It's rare to see an echo tail longer than that except on very high latency connections, or when the echo is actually acoustically generated by the far end and not by network effects. I haven't done any real measurement on it, but I believe it's actually longer than 128ms. As I go higher and higher with echocancel values, the echo does get better, but is never totally eliminated. At echocancel=1024, there is still rather pronounced echo on calls in the local exchanges. The calls are also more or less half-duplex at that point because the vpm is filtering out so much of the signal as echo. I may just tell the client to look at a partial PRI. All this echo chasing is getting costly for them. Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Asterisk Clustering
Please, does anybody have a good document describes well the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers. Documentation?!... well... there's not much. It depends on what you're trying to achieve with your cluster. If you want a simple active/passive failover cluster, I'd suggest heartbeat/pacemaker for clusterizing the services coupled with drbd for replicating files. I recently set up a cluster like this that's now in production. This particular system connects to the PSTN via PRIs, and a specialized piece of hardware detects which system is the active node and physically routes the PRIs to that node. I should probably write something up and post it somewhere, but time is always an issue. If you need specific help with this kind of setup, though, feel free to ask, and I may be able to assist. If you want an active/active setup, I think you'll have to look into using dundi. - Noah ___ -- 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] Sangoma Wanpipe Driver Compile for DAHDI Failure
[14177.069426] dahdi: Version: 2.2.0-rc2 Are you sure you're using the latest stable release of Dahdi and not the rc? - Noah ___ -- 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] Asterisk-beginner : cannot make phonecalls usingAsterisk
Let's just simplify this a LOT: Your phones have no dialtone. This means they are not registering with asterisk. I see in your sip.conf, for both you phones, you have: host=X.X.X.X If you specify an address here, your phones will not register. Instead, to make your phones register, set it to: host=dynamic (It does not matter if the phones are configured dynamically via DHCP or statically configured, but they do need to be configured to try and register to asterisk) For both phones, you might also want to add: qualify=yes This will monitor whether or not the phones are in contact with asterisk. You can view this with sip show peers. Once you've gotten the phones to register with asterisk, THEN try having them call one another. - Noah ___ -- 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] IMAP Voicemail - can't get messages. Arrgh!
Hi -
I just deployed a system using IMAP Voicemail. During my testing,
voicemail worked fine. I could check vm from the phone, and the
messages would get marked as read, or I could read the messages in a
mail client, and the phone's mwi light would turn off. Very neat.
I'm not exactly sure when things got munged up, but something broke.
I can record messages with Voicemail(), but now when I access an IMAP
mailbox using VoicemailMain(), it always says there are no messages,
even when there clearly are (unread) messages in the IMAP mailbox.
I've also got the asterisk GUI running on this system, and its status
page (retrieved via manager, I believe, or maybe voicemail show
users) shows the correct message counts. I tried debugging manager
messages to see how it was getting the message counts, but I didn't
get any useful output. Does anyone know a better way (any way) to
debug issues with IMAP Voicemail? I do see an error on the CLI:
ERROR[20010]: app_voicemail.c:2026 mm_log: IMAP Error: Quota not
available on this IMAP server
Here's some background info:
Asterisk: 1.6.0.8
IMAP Server: dovecot 1.0.7
c-client: UW imap2007e
Config Files:
voicemail.conf
[general]
format = wav49
serveremail = aster...@rosecompanies.com
fromstring = ${VM_CALLERID}
emailsubject = New voicemail. Length: ${VM_DUR}
emailbody = ${VM_NAME}:\n\nYou have a new voicemail message. You
currently have ${VM_MSGNUM} messages in your
Inbox.\n\nFrom:\t\t${VM_CALLERID}\$
maxsecs = 600
minsecs = 4
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 20
userscontext = default
imapserver = localhost
imapfolder = INBOX
authuser = asterisk
authpassword = xxx
maxgreet = 360
operator = yes
maxmessage = 300
minmessage = 4
saycid = no
sayduration = no
envelope = no
review = yes
users.conf (a typical user):
[02]
username = 02
transfer = yes
mailbox = 02
call-limit = 100
fullname = Test User
cid_number = 02
context = DLPN_MainUsers
hasvoicemail = yes
vmsecret = xxx
email =
imapuser = allison
hassip = yes
hasiax = no
host = dynamic
nat = no
hasdirectory = yes
dtmfmode = rfc2833
threewaycalling = no
callwaiting = no
hasmanager = no
hasagent = no
canreinvite = no
insecure = no
pickupgroup =
autoprov = yes
label = 02
macaddress = 0004f200
linenumber =
LINEKEYS = 1
secret = xxx
disallow = all
extensions.conf:
exten = 000,1,VoicemailMain(s${CALLERID(num)}...@default)
Thanks!
Noah
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Re: [asterisk-users] IMAP Voicemail - can't get messages. Arrgh!
I'm not exactly sure when things got munged up, but something broke. I can record messages with Voicemail(), but now when I access an IMAP mailbox using VoicemailMain(), it always says there are no messages, even when there clearly are (unread) messages in the IMAP mailbox. This appears to be the same issue as was resolved in bug 14685. If you use the latest version of Asterisk 1.6.0 branch then you shouldn't have that issue anymore. Aha! Thanks, Leif! I'm not insane. OK, well, maybe I am. I didn't find that bug. I think I'm going to bite the bullet and go with 1.6.1.0-rc4. Some of those items in the 1.6.1.0rc4 changelog just look to good to be passed up (or too scary to ignore). Thanks, Noah ___ -- 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] FXS Line Voltage When Dahdi/Zaptel is off?
You mean when the driver is not loaded ? It doesn't. The driver enables the current drawn. Well that is my guess. But since I have one card handy I'll confirm for you. CONFIRMED. No power without the driver loaded Excellent. Thanks, Martin! I didn't have one to test with (yet). Martin On Thu, Apr 2, 2009 at 4:38 PM, Noah Miller noahisaacmil...@gmail.com wrote: Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah ___ -- 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 ___ -- 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] FXS Line Voltage When Dahdi/Zaptel is off?
Hi - Does anybody know if an FXS generates line voltage when Dahdi/Zaptel is disabled? Thanks, Noah ___ -- 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] Asterisk 1.2.31.1, 1.4.22.2, 1.4.23.1, and 1.6.0.5 released
The policy that we have been following is that only final releases will be announced to the asterisk-announce list. Betas and release candidates are not. The rationale is that asterisk-announce is supposed to be a low-volume list and that most subscribers to it would not appreciate all the noise of announcing release candidates or betas there. I should think that the policy could be amended; however, I'm not really in a position to make that call, nor do I know if you're a vocal minority or if most subscribers to the -announce list would appreciate seeing such messages. Survey? I would appreciate such postings to the -announce list. Even with the rc release notices, it will still be a very low volume list. - Noah ___ -- 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] blind transfer on hook-flash from SIP phone
Hi Marcelo - Is there any alternative to invoke mid-call services without using the # and * signals? I was expecting to use Hook-Flash either via INFO or RTP telephone-event. You can change the keys used to invoke a service in features.conf. I know many people use ## or #1 for blind transfer so as to avoid issues with IVR systems. You may also want to change who has access to the various features (caller vs callee). You can do this with flags in your Dial statement. For example, if you have set the 'T' flag, the caller can do a blind transfer. If you only have the 't' flag set (notice lower case) only the person receiving the call can do a blind transfer. - Noah ___ -- 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] Asterisk 1.6 dahdi only?
Hi Steve - New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. Welcome to Open Source! Seriously, look at the README files accompanying asterisk, dahdi, and libpri. They will give you compilation/installation instructions. You can also search this list with google: Search term site:lists.digium.com Someone take notice! we need a link to instructions right of the main asterisk page. If you have a need for documentation, you're more than welcome to write it (once you've figured out how to install asterisk). We all contribute however we're able. Well, some of us do. Now to answer your questions: My 1st question is am I missing a good step-by step for 1.6 and how to compile/install it along with it's side components (dahdi/libpri)? when/if those side components are actually needed? When would you run asterisk without them entirely? 2nd question is for an IP/SIP only system do I only need DAHDI or do I need DAHDI and LIBPRI? If you have no dahdi compatible hardware, you don't need dahdi. The one exception to this is meetme, for which you need a dahdi timing source. You can use the dummy timing driver. Is libpri only needed if interfacing to a pri? Yes, mostly. I think you may need it if you have any card that takes a T1/E1. I think you may also need it for BRI cards. Is 1.6 so cutting edge that I should not expect to find complete documentation (yet)like I seem to be expecting very easily? The short answer is yes, given the glacial pace of documentation creation, 1.6 is that cutting edge. - Noah ___ -- 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] RFC -- Improving the quality of the mailing lists
It seems to me that everything one may want to know would be contained on voip-info.org Hmm. Dangerous statement. There are many things on the WIKI that are quite outdated, and a great many other things that aren't there at all. People don't ask stupid questions because of a lack of a FAQ to read, they ask stupid questions because they're too lazy do to the footwork. True. They may not know how to look up the answers to the stupid questions, though. I think a FAQ would help greatly in these cases. - Noah Robert Broyles wrote: I think we'd be better off posting a regular FAQ, perhaps weekly, with some of these suggestions, as well as providing a link to that FAQ from the mailing list signup page, along with a STRONG suggestion to peruse the FAQ first. I agree with this 100% I'm still pretty new to the mailing lists myself. I don't consider myself a novice Asterisk user, but one of my biggest 'complaints' is the lack of a well documented FAQ or Manual for Asterisk. (Unless one is willing to buy or read O'Reilly's Book - http://www.asteriskdocs.org - which quickly will be outdated again.) I have made it a personal aim to document all my findings in a blog, so that it's at least searchable by others through Google, in hopes that others might find it useful. But if we had a REGULARLY updated FAQ/Manual ... I think that would greatly cut down on the clutter posts. ___ -- 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 ___ -- 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] SLA and Polycom
I don't believe that Polycom's version of SLA does anything with Asterisk. You have to use asterisk's SLA implementation (http://www.asterisk.org/node/48342). So asterisk can't do SLA with Polycom phones? Asterisk can do SLA with Polycom, just not using Polycom's SLA implementation (in other words, don't bother setting up shared lines in the polycom cfg files - it won't do anything). You use asterisk's SLA implementation. - Noah ___ -- 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] SLA and Polycom
Hi Mark - Has anyone done SLA with Polycom phones? I've got a large project coming up where the customer is keen on SLA for trunks and extensions. Trunks will be on a PRI. We may do this with Cisco phones if they work better. You really want to do SLA with all 23 lines of the PRI? That's a lotta lines to be shared. You'd need two sidecars for each phone (Cisco or Polycom). - Noah ___ -- 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] SLA and Polycom
Hi Mark - You really want to do SLA with all 23 lines of the PRI? That's a lotta lines to be shared. You'd need two sidecars for each phone (Cisco or Polycom). Actually there will be multiple PRI's :) This customer is a multi-tenant situation so each tenant will have a few trunk SLA's and maybe some extension SLA's. Aha. That makes more sense. This is, they will if a) it's do-able b) it works on Polycom as I don't see anything coming back from the phone when I designate a line key as shared. I don't believe that Polycom's version of SLA does anything with Asterisk. You have to use asterisk's SLA implementation (http://www.asterisk.org/node/48342). - Noah ___ -- 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] noise in Asterisk 1.4 and 1.6 versions
Hi Abel - I had installed Asterisk 1.4 and when I call to a exist extension, the voice have noise, but, when I call to a extension does no exist, asterisk played a voice that say me that extension does no exist, but without noise I want I some body can test with a softphone my server, ip: 75.74.115.209 user: ramses pass: ramses the extension 1000 exist, try what ever other extension does not exist to hear the difference.. I would be willing to bet that the clear voice that you hear is generated by your phone (probably x-lite?), and not by asterisk. I'd also be willing to bet that fuzzy voice is caused by a bug that is present in certain versions of gcc. What version do you have? You can fix by compiling with the DONT_OPTIMIZE option (which should give you clear sounds), or just upgrade to a more recent version of gcc. - Noah ___ -- 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] IMAP Voicemail and Directory not working?
Hi Tzafrir - I'm wondering if anybody has IMAP Voicemail AND the directory working together. I haven't had any success. IMAP voicemail works fine, but when it's active, the Directory does not work. The problem seems to be with libc-client. Specifically, asterisk is not able to access the mm_dlog function. I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu 8.10 and Fedora 9. In each case, I used the native package manager to install libc-client, and in each case, after asterisk is compiled and voicemail users are configured, I get an error in the log that says this: On Ubuntu and Debian (Lenny/Sid) - apt-get source asterisk # as root / using sudo: apt-get build-dep asterisk cd asterisk-1tabtab ASTERISK_NO_DOCS=yes fakeroot debian/rules build Does it build? If so, you have a similar version of Asterisk that builds with IMAP support. I finally got this to work. For some reason, none of the packaged versions of libc-client from any of the distributions I tried support mm_dlog, which is required by the Directory app. I ended up compiling from uw-imap's source on Ubuntu, and that worked right away. On the Red Hat varieties, compiling from source worked, but I had to specify -fPIC and a few other compiler flags when building UW's c-client. For the record, if anybody needs to do this on a redhat platform: 1. Download imap-2007e (or latest version) from ftp://ftp.cac.washington.edu/imap/ 2. Unpack and compile with a make command like: make platform SSLTYPE=none EXTRACFLAGS=-DIGNORE_LOCK_EACCES_ERRORS=1 \ -I/usr/include/openssl -fPIC -fno-strict-aliasing -Wall -Wno-pointer-sign -Wno-parentheses (See the Makefile for a list of platforms - I used 'lr5' for CentOS 5.2) 3. In the asterisk source, run the configure script with the imap flag: ./configure --with-imap=/path/to/imap-source (use the base directory of the imap source - e.g. /usr/src/imap-2007e ) 4. Run make menuselect for asterisk and select IMAP_STORAGE from the Voicemail Build Options. Of course, you'll also need an appropriately configured IMAP server (for CentOS, I recommend their default choice of Dovecot). - Noah ___ -- 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] IMAP Voicemail and Directory not working?
Hi All - I'm wondering if anybody has IMAP Voicemail AND the directory working together. I haven't had any success. IMAP voicemail works fine, but when it's active, the Directory does not work. The problem seems to be with libc-client. Specifically, asterisk is not able to access the mm_dlog function. I've tried with Asterisk 1.4.22+ and 1.6.0+ using CentOS 5.2, Ubuntu 8.10 and Fedora 9. In each case, I used the native package manager to install libc-client, and in each case, after asterisk is compiled and voicemail users are configured, I get an error in the log that says this: [Dec 22 15:19:15] WARNING[24536] loader.c: Error loading module 'app_directory.so': /usr/lib/libc-client.so.2007b: undefined symbol: mm_dlog I also tried compiling from UW's c-client source, and I can clearly see the mm-dlog function in the source, but when compiled and linked into the shared object library, asterisk can't seem to access it. Does anybody have this working? If so, how did you do it? Thanks, Noah ___ -- 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] app directory error: libc-client undefined symbol
Hi Sean - On Wed, Dec 3, 2008 at 7:36 PM, sean darcy seandar...@gmail.com wrote: Installing 1.4.23-rc2, I actually looked at the startup and saw this warning: WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module 'app_directory.so': /usr/lib/libc-client.so.2007: undefined symbol: mm_dlog I'm running Fedora Core 9, with libc-client 2007d. googling didn't help, so what's the problem? Do I need a more recent (different) libc-client? I've got the same problem here with CentOS 5.2 and Asterisk 1.6.0.3-rc1. I tried rebuilding the libc-client rpm's from the source rpm, but the problem is still there. I'm trying to build libc-client from UW's source, but it seems to be a non-trivial thing. I'll let you know. Thanks, Noah ___ -- 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] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Ah, what a tease! For the client that would want this, I'm going to be upgrading their Exchange 2003 cluster to 2007 in a few weeks. Oh well. Thanks for the info. - Noah ___ -- 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] IMAP voicemail with Exchange (was: A way to run extenrnotify when IMAP events take place...)
Hi Jeff - I have IMAP voicemail working with Exchange 2003 using a single username and password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes. Care to share how you accomplished this? Thanks, Noah ___ -- 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] Setting up to reveive faxes.
Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. - Noah ___ -- 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] Limit the number of users in a meetme conference?
Hi Dan - I found the maxusers defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. That feature is primarily used with RealTime conferences. The maxusers value is read from a database and enforced on RealTime enable conferences. This presumes you are looking at 1.6.X or Trunk code... Ah. No realtime for me, so I guess I'll just stick with using MeetmeCount() in the dialplan. Thanks for the info! - Noah ___ -- 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Due diligence is required on anything 10,000 people are going to be pounding on. Undersizing is common, I think due diligence is THE key with any open source solution, including asterisk. I'll admit that I pretty badly screwed up one asterisk installation because I didn't adequately prepare it (shipped it to the customer and had their IT staff install - bad plan). And while I've never done a system anywhere near 10K extensions, I've had good experiences with some large-ish installations because I budgeted in the time for research and testing. I know that in the past there have been people on this list who have done very large scale asterisk deployments. Not sure if any of them are still around to comment. With that many extensions, I'll second using a SIP registrar like Freeswitch or OpenSer. Just use asterisk to provide the services. and is only one of the roads that leads to Hell (I prefer Patterson Lake Road myself since I drive in from the North East). Hmm. You must live near Ann Arbor. - Noah ___ -- 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] Large Asterisk installarions (~10, 000 extensions), preferably at universities
Is Asterisk even needed? Potentially, no. But if you intend to provide subscriber/PBX features, it is needed as a UA feature box(s). And FreeSWITCH can't handle that? Freeswitch can provide many PBX features with additional modules, but asterisk can provide more, and its implementations of such items are more time tested. One of freeswitch's big strengths is its ability to handle many SIP registrations. This is not asterisk's strength (at least not historically). One of Asterisk's big strengths is its multitude of services and features. This is not freeswitch's strength. Combine freeswitch and asterisk to get the best of both worlds. - Noah ___ -- 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] Limit the number of users in a meetme conference?
Hi - I found the maxusers defined in meetme.c, but I'm not sure how this value is set. Does anybody know if one can limit the number of users permitted in a meetme conference? I know there's MeetmeCount(), but I'd rather avoid the dialplan logic and just set maxusers instead. Thanks, Noah ___ -- 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] TDM2400P Voice Quality Problem
Hi Shariq - I m facing problem with TDM2400P pstn card. When someone dials, the voice quality is crappyInstead of hearing. Echo cancel almost works, but the callee hear what they describe as a 'background crackle/buzz' coming back when they talk. Crackling noise is usually caused by an unbalanced hybrid or a shared IRQ. Have you used the fxotune tool? This is the first thing you should do with any analog card. If you still have issues after running fxotune, check to see if your card is sharing interrupts with anything else like a network card or disk controller. You can use lspci -vv or cat /proc/interrupts for this. - Noah ___ -- 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] Asterisk connected to the PSTN vs. a commercial solution
Hi Alejandro - Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users and it works very well only in an intranet environment (no connections to the PSTN world). But in the near future, we have to plan a telephone system that works in the intranet (voip) and also it must be connected to the PSTN public network with a T1/E1 trunk, with 200 SIP users aproximately. So at first I have to ways to do that: 1- Continue using Asterisk and adding a T1/E1 interface in order to connect to the PSTN This is exactly what asterisk was designed to do. 2- Discard Asterisk and buy a commercial solution, because we have the money My questions are: does Asterisk work in the scenario I've described Yes. I've used it in just the way you describe in a number of production environments with great success. What is the best solution you can recommend to me ??? Get what you WANT. Both Asterisk and commercial solutions will probably work well for you (just be sure to use quality hardware). With asterisk you get great flexibility and expandability. With a commercial solution you get less of that, but you get to blame someone else if the system fails. Talk to management. What do THEY want? As has been discussed here before, nobody ever got fired for buying Cisco, but that doesn't mean Cisco is any better than any other vendor, including Digium/Asterisk. Find out what the needs of your company are and get the system that best fits those needs. - Noah ___ -- 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] Call transfer over IAX trunk
Hi Andrea - I have two asterisk servers, an IAX trunk between and some SIP users registered to each server. The scenario is this: user A, registered to PBX 1, calls user B, registered to PBX 2. Then A wants to transfer the call using the features.conf method (in my case, **), but is unable to do this. What flags do you have in your Dial() statement? If you want both parties to be able to transfer with the features.conf transfer, you need to have 'Tt' in your dial statement, like this: Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt) - Noah ___ -- 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] Asterisk connected to the PSTN vs. a commercial solution
Asterisks greatest strength is that it's a highly flexible platform that let's you pretty much do anything. It's downside, is that it's a highly flexible platform that let's you pretty much do anything. In other words, the quality of what you are trying to do depends on the quality and volume of the development and testing. That's one of the best statements about deploying asterisk that I've yet read. 1) Research Research Research 2) Plan Plan Plan 3) Build/Implement 4) Test Test Test Test 5) Deploy If you don't feel like doing steps 1, 2, and 4, then go with a commercial solution where they've already done those things for you. You'll likely sacrifice flexibility, but those things are taken care of (or should be) by the vendor. - Noah ___ -- 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] The problem DIAL with option T,t
Hi Larry - This is my setup of the features.conf but it had not any reaction after I pushed the *2 while calling was acting ! Could you tell me the reason ? Or give my the method of the setting. Thanks! LARRY [general] parkext = 700 parkpos = 701-702 context = parkedcalls [featuremap] atxfer = *2 [applicationmap] set(DYNAMIC_FEATURES=tranf) tranf = *2,peer,waitexten(10|m) You've got a few problems here: 1) You have two different operations set to: *2 You can only have one feature per key combination 2) You can't set the DYNAMIC_FEATURES variable in the features.conf file. You can only set variables in extensions.conf (or extensions.ael) 3) If you just need to set up attended transfer, you only need the line atxfer = *2 and nothing else. Attended transfer is a pre-defined feature. The [applicationmap] section is for creating new features that aren't pre-defined. - Noah ___ -- 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] Multiple Asterisk SIP Server/client connections
Hi Ken - The SIP.CONF has been made identical across all 3 remote locations, and the main server has the same config for each remote site connecting. I first want to confirm that it's possible to have 3 remote Asterisk servers setup as a SIP client connected to a 4th Asterisk server. I just want to double-check the setup you have: you say the main server has the same config for each remote site connecting. Does that mean they're all connecting to the same SIP user/friend account? If so, that wouldn't work. You need to have a unique SIP account for each SIP device that's connecting. If that's not the case, and you have a unique sip account for each of your Polycom devices, can you show us the relevant part of your sip.conf from the main asterisk server? Also, do you get any particular messages on the console regarding this? Have you tried turning on SIP debugging? Thanks, Noah ___ -- 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] sometimes extensions can't be called
Hi Nhadie - Could it be my problem is since i'm using 2 asterisk, if an extensions registers on asterisk#1 it will not be reachable by extensions on asterisk#2? or it should not matter if i'm using realtime? It does not matter that you're using realtime. If a phone registers to asterisk server #1, and another phone registers to asterisk server #2 they will not be able to contact each other unless the asterisk servers are correctly configured in a dundi cluster, of if you have explicitly configured sip or iax connections between the servers. I would suggest that you leave your configuration as is, but change the dns records for your asterisk servers to SRV records with different priority values. This will prevent phones from registering to both servers at once. The phones will only register to the asterisk server with the lowest available priority value. Note: this type of setup will act as an active-passive failover cluster. If you want an active-active load balancing cluster, you should look at using dundi. - Noah coz this is what i noticed: i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 asterisk #2 i thnk cannot find 118102 because it is registered on asterisk#1? hope you can enlighten me on this. thank you. regards, nhadie Darryl Dunkin wrote: Try setting 'qualify=yes' in the sip.conf for the users. This will send a SIP options packet every two to the phone to verify the remote NAT device is allowing traffic from both sources to the phone. Afterwards, you'll usually see this status from the servers, to verify the phone is reachable: 123/12364.23.49.5 D N 15103OK (44 ms) If one server is unable to reach the phone, the status will instead be 'UNREACHABLE'. If it is a NAT device with a stateful firewall, it will likely only open the port for one source IP, and not both servers. Issues like this are why I run in an active/standby setup as opposed to active/active. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Wednesday, July 23, 2008 03:40 *To:* asterisk-users@lists.digium.com *Subject:* Re: [asterisk-users] sometimes extensions can't be called Hi, I think i notice the problem now, but unfortunately i don't know how to fix it. i'm using 118103 i dial 113102 i got this on asterisk server #1. [Jul 23 18:27:48] -- Called 118102 [Jul 23 18:27:49] -- SIP/118102-08237ef0 is ringing what i did is keep on dialing then hang up dial then hang up, until i notice that when i dialed it went to asterisk #2 on asterisk 2 i see this: [Jul 23 18:30:40] -- Called 118102 but no ringing, it seems like it's trying to look for it, could it be because 102 is registered only on asterisk #1? but if i execute sip show peers i can see 118102 on both servers. i also had the problem wherein after i dial 118102, it goes to asterisk #2 and cince there is no ring, i hang up my phone, then i dialed again this time i see: [Jul 23 18:32:47] ERROR[17368]: chan_sip.c:3269 update_call_counter: Call to peer '118102' rejected due to usage limit of 2 yup i did set the limit to 2 but there was no asnwer on 118102 and i hangup, why did i reached the limit? Thanks in advanced Regards nhadie --- On *Wed, 7/23/08, Darryl Dunkin /[EMAIL PROTECTED]/* wrote: From: Darryl Dunkin [EMAIL PROTECTED] Subject: RE: [asterisk-users] sometimes extensions can't be called To: [EMAIL PROTECTED], asterisk-users@lists.digium.com Date: Wednesday, July 23, 2008, 5:13 AM Are the users registered to both active servers? 'sip show peers' in the console should make this obvious. If users are to call each other, they both need to be registered to the same server, or their client needs to be configured to register to both. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Nhadie Ramos *Sent:* Tuesday, July 22, 2008 21:52 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] sometimes extensions can't be called Hi All, I have 2 asterisk servers connecting to a mysql cluster. I'm using realtime on both asterisk. users register via domain, i have that domain on round-robin. users can register and sometimes can call each other, but sometimes even if an extension is register and i tried calling it, i got this on the the cli: [Jul 23 12:44:52] WARNING[32259]: app_dial.c:1183 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Jul 23 12:44:52] == Everyone is busy/congested at this time (1:0/0/1) but xlite or ip phone shows the extension is registered. but asterisk says it's busy. phones are
Re: [asterisk-users] 3-way calling for IAX channels
Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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] 3-way calling for IAX channels
Hi Daniel - There is no way to enable it at the softphone itself? As is the case for hardphones like my Polycom. A phone can definitely do conference mixing. As you asked about IAX channels on the asterisk-users list, I assumed you were asking about how to do this in asterisk. My experience with IAX softphones is somewhat limited, but maybe if you indicate which phone you're using, somebody could provide you with assistance. - Noah Daniel On Tue, Jul 22, 2008 at 8:18 PM, Noah Miller [EMAIL PROTECTED] wrote: Hi Daniel - How can I made a 3-way conference betwwen IAX channels? My current version is: 1.4.21.1 Anytime you need a call with more than 2 parties, you need to use some kind of conferencing application. The default conference application for asterisk is meetme. You can use meetme with any kind of channels (IAX, SIP, MGCP, ZAP/Dahdi, etc). Just use the meetme() application in extensions.conf, and create your conference rooms in meetme.conf - Noah ___ -- 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 ___ -- 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] Help With dial plan
Hi James -
Thanks for the wild guess. But The user(who is myself) is dialing 3000. It
only failes to work when I use patterns. So I thought I am making a mistake
on the syntax, I have checked all the books I have and the internet and I
can't see anything wrong. :-\
Sounds like time for some more in depth troubleshooting. What happens
when you follow Mark's suggestion of adding a NoOp statement? What
happens when you create other pattern-match extensions? Do they work?
What messages are you getting on the console? Is the call being
rejected by the SIP device? What messages do you get when SIP
debugging is turned on? etc, blah, blah, blah...
- Noah
Rizwan Hisham wrote:
maybe the user is dialing something other than 3000 and that extension is
not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED] wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be doing
wrong?
James
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Best Regards
Rizwan Hisham
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Re: [asterisk-users] Echo Issue
Hi Joseph - I have Astra 480i's and Snom M3's. I am using a SIP provider so I do not have any peripheral cards. I am on voip-wiki now reading about the echo canceller tuning, thanks! For your particular case, you're probably not going to find much useful info on the wiki about echo cancellation. The info there is about reducing echo when there is an analog-to-digital conversion (in other words, if you're connecting to PSTN lines somewhere). If you have echo on calls that go through your SIP provider, it is possible that they are not doing a very good job with echo cancellation. If the echo is exclusively on these calls, you'll probably want to call them to discuss this. If you have echo on calls between your Astra and/or Snom handsets, you may want check the gain settings on these devices. Reducing the gain would probably lessen the effect of the echo. You may also want to check if either of these phones is doing any AEC (acoustic echo cancellation), and if there are any AEC parameters that are adjustable. I don't have experience with either of these phones, so I can't give you direct info on how to do this, but I'm sure that at least Snom support can help you. - Noah ___ -- 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] Echo Issue
This is almost standard with voip calls. The echo-cancellation has to train up to the call parameters. Some hardware is better with it than others and you can try tweaking the value for the echo canceler up and down. Hmm. This has not been my experience. I have rarely seen echo on pure SIP calls, but in all cases that I have, I've found that it is a regular acoustic echo caused by unusual gain settings on at least one end of the call. On Sat, Jul 19, 2008 at 11:41 AM, Joseph L. Casale [EMAIL PROTECTED] wrote: I am being told by the users on a purely sip based setup that when an inbound sip call is first answered, they here an echo on their greeting and then the conversation stabilizes and it works well. Any ideas where to look to start curing this? Thanks! jlc ___ -- 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 -- Chad Whitten Metro Network Solutions (601) 366-6630 Phone (601) 366-6066 Fax (601) 842-6804 Cellular [EMAIL PROTECTED] ___ -- 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] Beep on transfer
Hi John - I have a request that I have not been able to figure out as yet. I need to be able to play a beep when a call is transfered via attended transfer. This is exactly what is in the bug tracker at: http://bugs.digium.com/view.php?id=3819 Has any one found a way, elegant ot otherwise, to make something such as this work? Thanks in advance for any help. Here's an incredibly inelegant way: When an incoming call hits an extension, set a channel variable to a particular value. Put a check in your extension logic to see if that channel variable is set (put this before you set the channel variable). If the variable is set, play a beep. If it's not, don't play a beep. Extremely hackish, but it would fulfill the request. Yes, this would play a beep if a call was just blind transferred. It would also beep if the call was parked (and possibly picked up by the same person), but you could also hack some more to avoid this. - Noah ___ -- 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] Digium PRI and Echo cancellation
Hi Loic - According to that its using MG2. I think it will say MG2 regardless of whether or not there is a hardware module present. Shouldnt it be using something like HPEC? I don't think the hardware echo cancellers use the HPEC algorithm. As Eric and Matt have mentioned, dmseg will tell you if a hardware echo cancel module is being loaded. - Noah ___ -- 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] Beginner Issues
Hi John - That could be...I only have ports 5060 and 8088 open on the firewall. Should another port be open? If asterisk is inside a firewall/nat and the phone devices are on the other side, you need to also open port for the rtp audio stream. By default, this is UDP 1 - 2, but this range can be modified in rtp.conf The phone I am using are pstn phones connected to a 2 port Linksys PAP2. I made sure that I checked the NAT option under the user account and enabled NAT Keep Alive under the PAP2 management interface. I am using the G726-16 codec for transmission. Aha. You're using the GUI. In that case, the useful info will be in users.conf. - Noah ___ -- 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] zap not getting callerid any more
One thing I have noticed is that in the cases where the wildcard cannot determine the CID (i.e. because the rxgain is up around 10.5), I get this in my asterisk console: [Jul 15 08:04:09] NOTICE[26696]: chan_zap.c:6670 ss_thread: Got event 18 (Ring Begin)... It is odd that it would work one day and not the next. I'd have to say, though that I've seen that rxgain/txgain values beyond +-8 seem to yield unpredictable results in many areas, even if they do get you closer to 14844, and that's even on the cool new cards all the kids are using these days. And now the obligatory: YMMV - Noah ___ -- 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] Reinvites and SIP/RTP
Hi Adrian - When I use re-invite, does the Asterisk server stay in the SIP conversation, and just RTP traffic diverts, or does the SIP transfer away from the A*k server too ? I'm sure somebody will correct me if this is wrong, but I believe the signalling must stay with asterisk, as asterisk needs to know if it should provide any services for the call (music on hold, transfer, etc). - Noah ___ -- 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] can not receive calls through pri
Hi Uros - I have problem using Asterisk.I have isdn-pri and openvox d110p card in my computer.They are connected with RJ-45 (1,2,4,5 pins to the card and all pins to the isdn done by telco workers). I got green led on isdn which is sign that isdn is working and that is connected to openvox, right ? I compiled newest versions of libpri zaptel and asterisk and had no problems during that. When I started services I can not receive any calls.No indication that any call is coming to Asterisk.When I dial number (to my line coz it is IN service so they can only call me not other way) I can hear telco message then few seconds of silence and busy signal. On cli I can not see anything.By the way I use Fedora 9 x64 kernel (I tryed with i386 kernel, with different machines,different distributions too but same problem occurred. Just to double-check: did you use the patched wcte11xp.c file from the openvox website? - Noah ___ -- 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] (no subject)
Hi - I'm trying to install a fresh copy of asterisk on a 64bit platform. I'm using CentOs 5.1 and all the latest builds of zaptel, libpri and asterisk. When I try to build Asterisk this is the error I'm getting. src/add.c:1: error: CPU you selected does not support x86-64 instruction set You may not have the right sources for your kernel. You may have the 32-bit sources instead of the 64-bit ones. What kind of CPU is it? - Noah ___ -- 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] Beginner Issues
Hi John - I have a fresh copy of asterisk, libpri, zaptel, asterisk-addons, and asterisk-gui installed on centos (I built everything using ./configure, make, make install, make samples). I connected to the GUI interface and created two new users. I used the two users accounts to connect up a couple of IP phones for testing. The phones connect to the server just fine, and I can even place a phone call to the other phone. However, I cannot hear anything on the dialed phone. The only thing I am able to hear is my own voice looping back to the phone I place the call from. Any ideas as to what I am missing? Most probably it's a codec issue, but we'll need to see your sip.conf file. It might also be helpful to know what SIP devices you're using. - Noah ___ -- 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] Poor audio quality with TDM400 card
Hi Leotis - When i run fxotune -i i get the following output: sudo fxotune -i Tuning module /dev/zap/1 Done! /dev/zap/2 absent: No such device or address /dev/zap/3 absent: No such device or address /dev/zap/4 absent: No such device or address /dev/zap/5 absent: No such file or directory /dev/zap/6 absent: No such file or directory /dev/zap/7 absent: No such file or directory /dev/zap/8 absent: No such file or directory /dev/zap/9 absent: No such file or directory /dev/zap/10 absent: No such file or directory /dev/zap/11 absent: No such file or directory is this the expected output ? Yes, if you only have one Zap channel configured. If you specifically have problems playing back gsm files, make sure you're not dealing with the gsm playback bug. Basically, if you compiled with the default options using GCC 4.2, gsm transcoding may be distorted. See here: http://bugs.digium.com/view.php?id=11243 - Noah ___ -- 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] Asterisk behind NAT, Polycom behind NAT (SIP), how to work?
Hi Bilal - When Asterisk behind NAT and Polycom behind NAT, I forwarded the 5060 UDP to asterisk (at asterisk router) and to Polycom IP Phone at polycomg router site, but the problem stayed. Also I was use nat=yes in the sip.conf Also I forwarded the udp rtp ports (that configured in rtp.conf) to the asterisk IP address, and did not succeed. Only forward ports (UDP 5060 and RTP) at the asterisk end. Do not forward any ports at the phone end. - Noah ___ -- 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] Poor audio quality with TDM400 card
Hi Leotis - Now that you mention that, i didnt even know there was a gsm bug. I am using asterisk 1.4.21.1, i visited the link you gave. I am guessing i will have to patch my asterisk installation, i am reading, the bug report to see,how i can verify that i have the gsm bug. Well, if you have gcc version 4.2.x (you can check with gcc -v) there's a good chance this is the problem. If i do have the gsm bug,how can i fix it. You won't need to patch asterisk. The bug is actually in GCC. You have two options: 1) compile with GCC 4.1 instead of 4.2, or 2) compile with the DONT_OPTIMIZE flag. I'd probably pick option 1, but it may just be easier to use option 2 depending on what gcc packages are available for your system. - Noah On Mon, Jul 14, 2008 at 11:40 AM, Noah Miller [EMAIL PROTECTED] wrote: Hi Leotis - When i run fxotune -i i get the following output: sudo fxotune -i Tuning module /dev/zap/1 Done! /dev/zap/2 absent: No such device or address /dev/zap/3 absent: No such device or address /dev/zap/4 absent: No such device or address /dev/zap/5 absent: No such file or directory /dev/zap/6 absent: No such file or directory /dev/zap/7 absent: No such file or directory /dev/zap/8 absent: No such file or directory /dev/zap/9 absent: No such file or directory /dev/zap/10 absent: No such file or directory /dev/zap/11 absent: No such file or directory is this the expected output ? Yes, if you only have one Zap channel configured. If you specifically have problems playing back gsm files, make sure you're not dealing with the gsm playback bug. Basically, if you compiled with the default options using GCC 4.2, gsm transcoding may be distorted. See here: http://bugs.digium.com/view.php?id=11243 - Noah ___ -- 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 -- Leotis Buchanan Manager/Electronic Design Systems Engineer Exterbox.com ___ -- 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] How to integerate 2 TDM cards on same machine.
Hi Syed - I have been using single TDM800P card. It is a small card with 4FXO and 4FXS ports. I have been using it for sometime without any problem. I am using Asterisk 1.4.18.1. Now due to greater requirement to handle more calls our office has bought another larger card TDM2401E which has 24 FXO ports. I have installed it on the same machine. Would like to know following about its configuration. Same Zaptel Driver will be used which is catering for my TDM800P card?? My zaptel.conf has following current config: loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8. Just add in the extra channels: fxoks=9-32. Be sure to check the order the cards are loading with zttool. If the 2401E is loading first, it will actually be channels 1-24, and the 800 will be channels 25-32. Also, test to make sure your machine is capable of this setup. Each of these cards will generate 1000 interrupts per second. Most modern motherboards should be able to handle this, but some older ones may choke under this load. - Noah ___ -- 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] Zaptel problem with pots lines
Hi Enrico - I'm trying to get up and running a TDM400 with a standard italian pots line but i'm having problems at getting asterisk to detect when the line get answered on outgoing calls. I'm using asterisk 1.6 beta 9 with zaptel 1.4.11. Zaptel channels use fxs_ks signalling . I must admit I know nothing about Italian phone lines, but maybe you could try other signalling methods? Maybe ground start or loop start would work. - Noah ___ -- 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] AsteriskNow SIP config
Hi - I can not seem to get AsteriskNow to register my SIP provider correctly? I can do this manually when compiling Asterisk and installing it w/o a GUI, but not with this. I just get the following message. -- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #22) The register line I use normally looks like: user:[EMAIL PROTECTED]:port but the above looks simplified? Is that only a result of what the logging looks like? Any ideas? You can always edit the config files by hand. I had to do this on an AA50 I installed. The GUI mostly works, but if you need to fill in holes via CLI, you have that option. - Noah ___ -- 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] How to integerate 2 TDM cards on same machine.
Hi Syed - zttool shows that TDM800P is loaded first and TDM2401E is loaded second. now problem is ports are not being configured by asterisk. i have done following changes in two files zaptel.onf and zapata.conf. zaptel.conf loadzone=us, defaultzone=us, fxoks=1-4, fxsks=5-8, fxsks=9-32(or should this be fxoks???) zapata.conf signalling=fxoks channels =1-4 signalling=fxsks channels = 5-8 signalling=fxsks channels = 9-32 please see the bold lines. since FXO ports use FXS signalling so i used fxsks. is this right or wrong. are these changes have to be made in both the files as i have done or only in zaptel.conf waiting for information Almost there. Your zaptel.conf is correct (sorry I gave you the wrong signalling before). In zapata.conf, your signalling lines should look like: signalling=fxo_ks channels = 1-4 signalling=fxs_ks channels = 5-32 - Noah ___ -- 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] Incoming calls on zaptel not answered.
Hi Jose - After an upgrade from asterisk 1.2.13 to 1.2.25 including upgrading libpri, zaptel, the incoming calls to a TDM400P REV I, with 3 FXO modules stop working. The board is working, I tested in another server with the 1.2.13 asterisk version. Also changed the pci slot where the board is. Hmm. Bad or incompatible PCI slot? Can you (at least for testing purposes) switch back to the original PCI slot you were using when the card worked? - Noah ___ -- 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] Problem compiling Zaptel
Hi Bob - I have a problem compiling Zaptel on an up to date CentOS 5.2 box. Zaptel 1.4.11, CentOS running on AMD dual core X64. ... CC [M] /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o In file included from /projects/asterisk/zaptel-1.4.11/kernel/xpp/xpd.h:26, from /projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.c:27: /projects/asterisk/zaptel-1.4.11/kernel/xpp/xdefs.h:117: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here make[4]: *** [/projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 Are you using any Xorcom hardware? If not, you can avoid this issue by disabling the appropriate items when you run make menuselect before compiling Zaptel. If you are planning on using Xorcom hardware, there is a patch available, which I believe is on the Xorcom website. - Noah ___ -- 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] new install of asterisk appliance.
I have 1 nic card which is linked to the router. Then I use 1 port on the router which is linked to the asterisk appliance. It will work via WAN which ive now got. SO I can access the asterisk appliance via 192.168.1.15 The problem is now…How do I connect the phone. Ive got the phone (Ethernet) connected from the LAN port on the phone to a LAN port in the asterisk appliance. The short answer is that if you're not using the AA50 as a router you cannot use the LAN ports on the AA50. You'll have to connect your phone to a LAN port on your switch. As Rob mentioned, you'll need to configure the Grandstream manually. Also, if your SBS server or your router is providing DHCP be sure to turn off DHCP on the AA50. - Noah ___ -- 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] Choppy audio
Hi Doug - In my research it appears this often happens when using more than one processor. I am using a dual core Pentium. I guess my dilema here is which way to go. Clearly the audio is not working the way I would like it to and the way I came to expect from my old system. When playing messages it seems to get out of sync. Sometimes skipping ms's of audio. This seems to happen at about a 2-4 second rate. I believe that I have things setup to use the RTC as a timing device (see below) but that did not seem to change the problem. It may have made it better but not much. What are my choices? HW card?, Upgrade Asterisk?, The symptoms don't sound exactly the same, but is it possible that this is the GSM/GCC playback bug? http://bugs.digium.com/view.php?id=11243 - Noah ___ -- 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] fxotune vs rxgain/txgain
Hi Matt - In short, fxotune adjusts line impedance, where as adjusting gains I believe is essentially adjusting the amplification / deamplification of the signal. http://www.voip-info.org/wiki/view/Asterisk+fxotune Well, that clears it up a little. I think where I get confused is that sometimes using fxotune is called balancing the hybrid and some times using ztmonitor and adjusting the txgain/rgain settings is called balancing the hybrid. Perhaps they both try to achieve the same goal, but through different means? This leads me to my other question - Are these two techniques mutually exclusive? In some posts from Matthew Frederickson, it seems that they are, and that if you use fxotune, you should set your gains back to zero. Some other people seem to suggest using both fxotune and adjusting gain levels. I note that Stephen Bosch asked just this question some time back, and nobody was able to answer him. Does anybody know? Thanks, Noah ___ -- 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] fxotune vs rxgain/txgain
Hi Matthew - These techniques are not mutually exclusive, I usually want people to use gain modification as the last step in trying to eliminate echo (after balancing the hybrid and making sure you are using a good echo canceller). In the case of running fxotune, your zapata.conf software gain levels should not affect its operation. If you are using any of the hardware gain settings (wctdm24xxp module parameters) you should normalize those to 0 beforehand so that they do not interfere with the calibration process. Thanks for your responses! I actually didn't realize there are hardware gain settings available for wctdm24xxp (is there any documentation on this? I can't seem to find any). I assume the hardware gains default to 0 if left unset? Just two more questions: 1) I think we were experiencing ECFO with an rxgain setting of +10db (after having balanced the hybrid using fxotune). I'm guessing this is because that rxgain value amplifies the echo a bit too much. I know this is a bit of a loaded question, but is there a certain range of values for rxgain/txgain that we should stay within in order to avoid exacerbating any echo issues? 2) Are rxgain/txgain values applied before or after hardware echo cancellation? Thanks, Noah ___ -- 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] bad call quality
Hi Edd - I run a couple of asterisk servers all connecting to international sip providers. All three servers are on the same type of internet connection (Martis/Diginet). There isnt a shortage of bandwidth, and its not a codec issue, as ive tried swapping codecs. If its not a line issue, because if i route the calls via sip via another server(which i own)(in same country) and then break out from there i get good quality, but im paying for triple bandwidth then, and bandwidth in south Africa is hellishly expensive. The Physical hardware is not overloaded either. I have tried rebooting my equipment, and that changed nothing either. if i do a ping flood i get decent results(well, only about 10ms more than another perfectly working branch) What else could this Be? Im completely Dumbstruck. Is there any other non-VoIP traffic using the same internet connection as the asterisk server? If so, this could very well be a QoS issue. You can get some nasty sounding calls even on a very fat internet connection if there is no QoS. One of my clients has a 100mb fiber connection to the internet, and we had to really fine tune their Cisco routers in order to get usable VoIP calls to their branch offices. I've also seen internet connections that are just very poor, and no amount of internal QoS can fix this. What kind of routing equipment are you using? - Noah ___ -- 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] Asterisk not picking up incoming calls from TDM400P
Hi Drew - I really don't know anything about how phone lines work in Singapore, but maybe you could try using ground start signaling (fxsgs)? - Noah ___ -- 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] Similar extension numbers for multiple users
Hi Zeeshan - If you have multiple tenants using the same extensions range, you have two options: 1) have the tenants call each other via their PSTN numbers, and then dial the internal 1XX extension 2) assign a special prefix for each of the tenants to call each other. For example, tenant one has a prefix of 1, tenant 2 has a prefix of 2, tenant 3 has a prefix of 3, etc. If user from tenant 3 wants to call someone from tenant one, they would dial 11XX, and to dial someone in tenant 2, they would dial 21XX, etc. If your SIP phones support non-numeric dialing you could add letter suffixes like you had suggested, but not too many phones support this. Personally, I'd forego both options above and assign each tenant to a unique extension range: tenant 1 gets 1XX, tenant 2 gets 2XX, etc. - Noah On Thu, Jun 5, 2008 at 8:34 PM, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Currently my devices are set as follows: Devices --- [100] type=friend secret=42335432 qualify=yes port=5060 host=dynamic dial=SIP/100 context=user1 canreinvite=no accountcode=user1 I guess I can change it to 100a, 100b and so on for different users. But I would need help with a sample context for how to make them dial out and each other. -- Zeeshan A Zakaria ___ -- 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
[asterisk-users] fxotune vs rxgain/txgain
Hi All - I hope somebody can clarify for me what exactly fxotune does, and how it is related to gain settings. I've been reading what appears to be conflicting information from various sources. I've got a box with an AEX800 with 6 lines (from Qwest) running asterisk and zaptel versions 1.4.20.1 and 1.4.11 respectively. We've been experiencing some echo/quality issues on certain calls which seem to happen on all 6 of the lines. I manually calibrated the rxgain/txgain using ztmonitor and a milliwatt test line to the somewhat improbable levels of +10.0/-2.0 (about the same for all 6 lines). These settings yield acceptable call volumes, but echo and noise are problems. If I run fxotune, it gives me the following numbers: 1=10,0,0,0,0,0,0,0,0 2=12,0,0,0,0,0,0,0,0 3=12,0,0,0,0,0,0,0,0 4=10,0,0,0,0,0,0,0,0 5=10,0,0,0,0,0,0,0,0 6=10,0,0,0,0,0,0,0,0 Two questions here: 1) What do these numbers mean? Are they in any way related to either rxgain or txgain? 2) Am I supposed to set rxgain and txgain back to 0 if I use fxotune -s? If I do use these fxotune settings and set rxgain and txgain to zero, the volume on incoming zap calls is almost too low to be heard, but echo issues seem to be solved. Do I have to choose between 1) acceptable call volume with echo or 2) super-quiet call volume without echo? Should I petition Qwest to install a repeater? Thanks, Noah ___ -- 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] No DTMF on Sip Trunk?
Hi All - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. I've tried using inband, rfc2833 and auto, and none of them work. Maybe I'm missing something obvious? Here's my config: Asterisk1 (1.2.18): sip.conf [129trunk551] type=friend secret= username=129trunk551 host=xxx.xxx.xxx.xxx context=phones dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Asterisk2 (ABE revC): sip.conf [129trunk551] type=friend secret=*** username=129trunk551 host=yyy.yyy.yyy.yyy context=default dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Thanks, Noah ___ -- 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] Playing mp3-files – will it b e OK?
Hi Harry - 99% of all my users are calling from GSM phones, and my system basically just plays some sound files back. The PBX is connected to an ISDN-30 connection. Are there any modules for playing MP3 files, so I can use them with commands like Play() and Background()? See asterisk-addons for the mp3 module. And will it have any effect on the quality? The callers should hear the file at the codec-quality of the channel they're connecting on. So for your ISDN callers, that's probably ulaw or alaw, and for the internal phones, GSM. - Noah ___ -- 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] No DTMF on Sip Trunk?
Hi Jared - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set rfc2833compensate=yes in the peer or friend section of sip.conf on the Asterisk 1.4 box. Unfortunately, this didn't work. Maybe rfc2833compensate isn't available in ABE? I think this may require inband signalling anyway, as we'll require non-sip (zap) devices to be able to use these sip trunks and enter DTMF. Any other ideas? Thanks! Noah ___ -- 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] No DTMF on Sip Trunk?
For ABE support you really should contact Digium. BTW, there is no such thing as a sip trunk. It's a sip peer or connection or account. shrug Semantics. IAX connections between two asterisk boxes are often called IAX trunks. This is the same thing in SIP flavor./shrug Also, no offense against Digium support, but the list actually responds more quickly at this point. I think the Digium support staff are in a situation of high demand and short staffing. - Noah Noah Miller wrote: Hi Jared - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set rfc2833compensate=yes in the peer or friend section of sip.conf on the Asterisk 1.4 box. Unfortunately, this didn't work. Maybe rfc2833compensate isn't available in ABE? I think this may require inband signalling anyway, as we'll require non-sip (zap) devices to be able to use these sip trunks and enter DTMF. Any other ideas? Thanks! Noah ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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
