[Asterisk-Users] Wireless LANs and Asterisk
Has anyone had any experience with wireless LANs and Asterisk? We have and here are my impressions. We configured an Asterisk in the office as a precaution to see how it would work for our own retail customers. Our office is open space, about 800 sq ft. (20x40 area). We use Snom200 and Grandstream SIP phones. Using the latest Linksys wireless access point (WAP54g) and 3 wireless bridges (WET54g), I have found that it works most of the time with WPA encryption on, but will occasionally drop voice (loosing packets). With no encryption on the WLAN it seems to work without a hitch! Using a less CPU intense encryption such as 64bit WEP, things also work fine. There must be too much delay with higher rate encryption. Also we had one bridge that seemed to be a week puppy in the litter. It could only muster 60-70% signal strength. It seemed to have problems under all configurations. Finally we positioned it such that it too works well running WEP 64b. I wonder if having 3 wireless bridges in close proximity would have anything to do with the signal strength? I would doubt it though. Anyone else with other experiences to share regarding wireless LANs and encryption? I'd me interested to hear them. Thanks, Mike Meyer GenDesign Corporation ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cannot call Grandstream
Michael Stephen; I have been running GS BT101's for the past few months in a fixed IP arrangement and have not had a problem with the registration process. Budgetones seem very reliable. I have the phones configured to do registration and expire every minute. I have SIP user ID and Authenticate ID set on the phone to the same as the phone number defined in the sip.conf file of course. Also no Authenticate password defined. I plan to move to a DHCP environment sometime soon. Hopefully I will be able to get through that too. Mike Meyer On Wed, 2004-10-20 at 12:57, [EMAIL PROTECTED] wrote: Michael, I have never been able to get the Grandstream to register reliably - with any version of the firmware. It sounds like in your test with the fixed IP, you left the registration option on the phone set to yes. With a fixed IP and host=IP address, I am pretty sure that you must turn off registration on the phone. It's useless anyway with fixed IP and just reduces reliability (as you have discovered the hard way). Asterisk periodically sends polling packets to the phone, so it will know when it is reachable and when it is not. And, the phone will still authenticate against the password, so this should not lower security at all. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Advice on OS Choice
On Fri, 2004-10-15 at 12:00, [EMAIL PROTECTED] wrote: Message: 1 Date: Fri, 15 Oct 2004 16:56:21 + (UTC) From: [EMAIL PROTECTED] (Tony Mountifield) Subject: [Asterisk-Users] Re: Advice on OS Choice To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] In article [EMAIL PROTECTED], Kevin Walsh [EMAIL PROTECTED] wrote: That's not up to them to decide. Under the GPL, if you distribute modified code then you must publish your enhancements for the benefit of all. The team responsible for the core code can decide whether the contributed code is appropriate for distribution. That's not how I understand the GPL. My understanding is that the GPL gives me the freedom to take some GPLed code, modify it, and distribute the modified code to whomsoever I choose, for free or for a payment. I must also make the source code freely available (on request, if I prefer) to anyone to whom I distribute binaries. It does *not* compel me to distribute my version to everyone, but I also cannot prevent those people I give or sell it to from passing it on in binary and/or source form to anyone else if they choose to. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org I think you are both right. As I understand it, it all depends on what is modified. Say in the case of Asterisk you modified and improved the conferencing capability, or voicemail, those modifications are made to existing features and are expected to go back into the GPL source. Final judgement is made by the GPL core distribution team to incorporate or not. Should you take and build a capability like Billing or another conferencing capability, or call transfer to your liking on top of the asterisk system. This in not required to be reviewed or submitted to become open source. It is your own software capability to do with what you want. Keep it proprietary or open source it. Your choice. Please correct me if I am wrong. Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 3, Issue 182
Sudhir, I have SNOM 200's. I agree the physical design is poor and have the same problem with finding the cradle and keeping the handset in the cradle. As for registration, I have my snom's set to register every minute per the settings on the phone. In this way it tries to register quicker after I change the settings. You might look to see what is being sent between the phone and the * SIP server. Turn on debugging from the CLI via sip debug peer x, where x is the phone extension. I also had problems after I had upgraded the firmware on the SNOMs. Sometimes there are new fields and settings that need to be filled in. Mike Meyer -- On Thu, 2004-10-14 at 04:24, [EMAIL PROTECTED] wrote: I purchased couple of SNOM 190 phones last week. Connected them to the Asterisk server, and they seemed to work fine. However, after sometime they seem to lose registration with Asterisk as I can make calls but cannot receive calls. I hope this is due to some incorrect setting on my part but cannot figure out what? Any suggestions? The headset (has Lucent's logo on them but look like Plantronics') would not work properly either. There is no audio on the other side whereas the same headset works great with GS HandyTone-Analog Phone combination. Finally, I tried putting an amplifier in the middle and now the headset works ok. The handset keeps falling off the cradle. Very poor design IMO. In any case, now that I have purchased the SNOM 190 phones (with a lot of expectation I must add), I have to live with it. Any suggestion about what do I need to change so that it can always receive calls. Thanks, -- sudhir ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: About 3 Way Calling on GS BT100
Leah, BT101's do not do conferencing yet. I was told by their support that this capability will be provided by the end of this year. Even though there is a conference key it doesn't conference. I just tested it after I saw your message and since I had loaded the latest firmware (1.0.5.14) into the phones earlier this week. It still doesn't support conferencing. Transfer works though, but that doesn't buy you much. What we do here is set up conference rooms and transfer callers into conference rooms, then call the conference room directly to join the conversation. A little cumbersome, but functional. Mike Meyer On Thu, 2004-10-14 at 13:00, [EMAIL PROTECTED] wrote: -- Message: 10 Date: Thu, 14 Oct 2004 13:44:01 -0400 From: Leah Newmark [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: About 3 Way Calling on GS BT100 To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 When I was researching this a bit, I thought I came across a tidbit that said that this phone is not capable of using flash...but now I can't seem to find that now, and on the contrary, the things I read say that it has such capability...Right now I am using a regular phone just with a SIP adaptor, and in order to accomplish 3 way calling, I hit flash, as I would if it wasn't using VOIP capabilities. But by reading http://www.grandstream.com/ user_manuals/budgetone100.pdf, it looks like that pressing flash does call-waiting, not a conference call...it might help to look that up a bit, as well as http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone% 20grandstream%20budgetone#comments ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Dial group continues to ring after answer - SNOM phones and solution
Asterisk Users;
Just wanted to let you know I fixed my problem.
To follow up on my own testing of the situation, I find that the
continued ringing after pickup only occured on the SNOM phones in the
group. The Grandstream phones stop ringing when another phone picks up.
Having turned on SIP debugging I have verified that the cancel message
is sent to the SNOM phone (and others in the group) when one of the
group phones is picked up, as expected.
It appears that the SNOMs don't handle the cancel message the same as
the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the
latest official release.
It seems that these phones even though they are set to do automatic
update, they do not. Or perhaps that capability was broken in the
firmware version I had last updated to.
THE SOLUTION:
To remedy the problem I upgraded to version 3.52 beta version. Also
2.04g fixes this problem as well.
I had to create my own internal TFTP server and flash update to 3.52.
The standard update process did not work to go beyond 2.03y or 2.04g. I
tried 2.05e f and these would never come out of boot.
MORAL TO THE STORY: Keep your SIP phone firmware up to date.
SNOM support is telling me to upgrade to 3.54. I don't see this one
listed on the standard update URL. I am a little leery about moving to
that one.
Now to upgrade my GrandStream's. They seem to be stuck at an old version
as well.
Thanks,
Mike Meyer
On Tue, 2004-10-05 at 16:47, Mike Meyer wrote:
Asterisk Users:
We have our * dial plan set up to ring 5 phones in the office and it
delivers the call to the first that picks up their receiver.
The problem is that after the pickup, everyone else's SIP phone keeps
ringing at least once and sometimes twice. This interferes with the
conversation, while others pick up the phone and get nothing.
Does anyone else have similar problems or have a solution to stop the
ring once answered? My dial statement looks like the following and has a
timeout of 15 seconds.
exten = MainTeam,1,Dial(${MainTeamChannels},15,tr)
exten = MainTeam,2,Voicemail(u${MainTeam_EXT})
...
note the variables MainTeamChannels define the SIP phone channels
defined in sip.conf and MainTeam_EXT is the voicemail box for this group
extension.
As an alternative, I am considering doing a round robin on a call group
or pickup group and implementing call pickup.
Any ideas welcome.
Mike Meyer
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[Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution
Someone pointed me here http://www.snom.com/downloads/share ... Yup! That where the SNOM support team sent me. Seems that they may be suggesting a different process or URL do update from. My concern is whether the latest version 3.54 has been tested and is an official release. I hate to put something out that hasn't been through a sufficient QA process. I don't want to risk getting my user's mad at me with a bad version of software. Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Random Intermittent Noise for SIP to FX0 calls plus echo
Dear group, Was wondering if anyone out there has had the experience I have been having. In reading recent posts on echo cancellation, I think there is We recently cut over the Asterisk and are configured with 5 FXS and 2 FXO ports to the PSTN via 2 TDM400P's and 5 SIP phones on our local network. I have set up echo cancellation with 800ms echo training. I do not have echocancelwhenbridged on, since since this is for complete TDM circuit per the comments in zapata.conf. When calls come in, there is echo, but it quickly trains and goes away. This is not the problem though. The asterisk server has 1GB RAM and 1GHz clock. We are currently using u-law codecs only. Digium support said that I might want to play around with using other codecs and the RX/TX gain to see if that makes a difference. These gain settings are not set and therefore taking the default. Before I start shooting in the dark, I thought I'd go to this group to see if the following problem has been solved before I start shooting in the dark. The problem we are having is that every now and then, say 4 times over a 20 minute call, interference occurs in the ear of the SIP phone user. The other side (PSTN caller) of the conversation may hear a few short (half second) breaks in the conversation. The characteristics of the interference in the SIP phone user is some in-band MF tones which may start out faint and get louder which are broken up them selves to make a cracking noise. There may be some momentary echo of my voice when this happens also. This may last for about 3 seconds and then the conversation is clear. We don't get this when using Analog phones on an FXS port nor for SIP to SIP conversations. Has anyone else experienced this and resolved it? Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Speech Recognition and Asterisk
All; Since I have interest in providing the capability for callers to speak the department, person or number they wish to call, as well as other IVR scenarios, I have been reviewing much of this lists email archives and searching the web for open source voice recognition that will work with the Asterisk PBX. What I am trying to determine, is what will it take to get it working on Asterisk? How much effort and cost? So far I have uncovered references to the following: 1) VoiceXML standards, and forums 2) OpenVXI - which supports VoiceXML, simulated speech, telephony 3) PublicVoiceXML 4) Sphinx - a Carnegie Mellon University Speech recognition project funded by DARPA From what I can tell, I feel I am uncovering the tip of the ice berg and this may not be trivial. But it seems that the Voice recognition application, once developed, would have to be linked via an AGI to the asterisk dial plan. Has anyone gotten Voice recognition working with Asterisk? Last I saw, a few were attempting to apply Sphinx back in the December and April time frame. Any shared successes, progress or direction on Sphinx or any other VR app would be appreciated before I start down this road. Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Distinctive Ring Cadences
Hello All, I am looking for a way to do priority call ringing. That is when a caller places a call to another party, they can indicate that the call is a priority and get a different ring to occur (ring cadence) on the called parties phone. This would be synonymous to an intercom ring on a key system. After some investigation, I have come across the ability of the GS BT101 which will ring differently based on the CID. But, this doesn't allow the caller to control the ring. I have uncovered some past discussions (http://lists.digium.com/pipermail/asterisk-users/2002-December/006378.html) and (http://lists.digium.com/pipermail/asterisk-dev/2003-June/000916.html ) regarding patches to support SIP phones and have no idea if they are implemented features or not. If anyone knows, please let me know. Thanks, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Directed Call Pickup
Hi everyone, I have been trying to get call pickup to work and am having success with group pickup by setting the callgroup and pickupgroup in the zapata.conf and sip.conf files. However, I cannot get directed call pickup to work. According to the little documentation (http://www.voip-info.org/tiki-index.php?page=PBX+Call+Pickup) that I found on the subject, the user is supposed to be able to enter the call pickup extension and then enter the extension to pick up. Basically *8 doesn't work. Is there some configuration that I am missing? Do I need to define a new call pickup extension in extensions.conf that answers the call on the remote extension? Any direction will help. FYI; Another caveat I have also run into is that a simple reload after changing the sip.conf and zapata.conf files did not always pick up the new group settings and allow me to do group call pickup. Anyone else run into this? I had to stop and start *. Thanks much, Mike Meyer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
