[Asterisk-Users] Asterisk/Metaswitch trunk, no inbound RTP stream on inbound calls
Ive been racking my brain for the last two days to try to figure out what I could possibly be doing wrong in my configuration for a SIP trunk thats setup through my local ISPs Metaswitch. Ive setup a very simple SIP Peer, which Ive played around with a lot in the past two days but still comes back to the following basic setup: [provider-fireball] type=friend insecure=very host=1.2.3.4 context=keysystem nat=yes canreinvite=no username=1235551212 fromuser=1235551212 secret=mysecret disallow=all allow=ulaw For whatever reason, on inbound calls, the RTP stream coming from the provider never initiates. The RTP stream on my side starts as soon as my dialplan is set to Answer() and we send a 200 OK back to the Metaswitch. However, after the 200 OK, we never receive an inbound RTP stream. There are no known configuration changes on their side that would cause this nor any configuration changes on my side. Its a very strange problem. Does anyone out there have any experience with interop between Asterisk and Metaswitch? More importantly, has anyone ever seen an issue where inbound SIP signaling works fine but no RTP inbound and its definitely not a firewall issues (verified with multiple packet traces before and after the firewall). Outbound calls work fine with the inbound and outbound RTP streams both good. I have plenty of packet traces available for people who are interested. Interestingly enough, a Linksys PAP2 on the same network works fine with the same switch. Any ideas? Clint ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP TAPI
FYI, I've got a working version of asttapi that will work with Asterisk 1.2 up on my site at http://www.kirkhamsystems.com/asttapi . It's the debug build, so it contains some extra code, but that's merely to help me out if anyone sends in a bug report (which so far out of apparently 80 something downloads, no bug reports yet, I guess it's working well). Only reason I mention it is that I can't imagine trying to drop down to SIP level support in asterisk when the asterisk management interface works so well with asttapi. Clint -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Wednesday, May 24, 2006 10:32 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP TAPI Hello, Anyone try to use SIP TAPI (http://www.enum.at/index.php?id=479) with Asterisk? Pretty nice, pretty simple. I am hung up on something, though, and google doesn't specifically address my issue. The program seems to go to the s extension in the default context of the sip user it is configured for. Is there a way to set it to go to an extension of the default context? I couldn't figure out how... Assuming you can't specify an extension within the default context, then that leads me to believe that a SIP user needs to be created specifically for each instance of SIP TAPI. So I tried that. I made a context just for this, with the s extension set to immediately dial my real sip phone. This works, and Asterisk bridges the call, but that's it. The docs are sparse, and I don't think this program was specifically written for Asterisk - what am I missing? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: SIP TAPI
Yeah, that sounds about right. I can see advantages and disadvantages to both. The main advantage I see to AstTapi besides signaling incoming calls (which I haven't tested on my modified code, I guess I should work on that) is that once you've setup a user in the Asterisk Management interface and modified your dial plan accordingly, you're done, you don't have to add new entries for every instance of AstTapi. That would be a burden I'd think in a larger installation of SIPTapi with Asterisk. The nice advantage also to AstTapi is that signaling is ongoing while the call is in progress, so you can end the call from the TAPI application. This is a real boon in real CTI setups for callcenters where the phones might be set to autoanswer incoming calls on a headset, display information, and the user ends the call. Seems like there should be a simpler way to do an TAPI interface with the Asterisk management interface w/o a bunch of UserEvents though. I think I'll look into that, because it'd be nice if all you had to do was add the user to the manager.conf and be done. I know I could probably do that on outbound calls, incoming calls might be a little more difficult. It could probably be done with some assumptions about extension length, etc. Sorry, just thinking aloud, but that's probably where it should go from here. Clint -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brent Torrenga Sent: Wednesday, May 24, 2006 12:44 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: SIP TAPI Clint, Crap. Wish I would have seen your setup first. I played with asttapi for a few days, and gave up. My problems were manager related, and you cover those points well enough on your page. I was able to get SIP TAPI to work this way: - each install of SIP TAPI needs a SIP user in sip.conf. - each SIP user made for SIP TAPI needs a context in extensions.conf. - each context made for SIP TAPI looks like: [blah-tapi] exten = s,1,Dial(SIP/blah) Include = blah-internal-context It seems to work great this way. The software is taken out of the loop immediately after connecting to SIP/blah, thus does not have call state like ast tapi does. However, I think this also means that you can have an unlimited number of simultaneous calls, unlike ast tapi. Also, this does not provide for pop-ups on incoming calls or call progress, whereas ast tapi does. What I really don't like about my setup is the lack of outbound caller-id on your phone - no way to use the redial button. I guess a plus for SIP TAPI here is that it doesn't require manager events to be put into the dial plan - yay! Clint, in your opinion, do I have the differences between the two programs summarized correctly? FYI, I've got a working version of asttapi that will work with Asterisk 1.2 up on my site at http://www.kirkhamsystems.com/asttapi . It's the debug build, so it contains some extra code, but that's merely to help me out if anyone sends in a bug report (which so far out of apparently 80 something downloads, no bug reports yet, I guess it's working well). Only reason I mention it is that I can't imagine trying to drop down to SIP level support in asterisk when the asterisk management interface works so well with asttapi. Clint Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Asttapi for Asterisk 1.2 Testers Needed (was RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial)
I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to test this out, I'll send you a zipped copy of the TSP file (I haven't worked on doing an installer yet). I need to send out the debug build so I can generate information in case it doesn't work on anyone else's PC. Contact me off-list for copies. Clint -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Friday, May 12, 2006 1:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Oh.. :/ too bad.. I'll have to look at the source.. bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw Sent: Thursday, May 11, 2006 11:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Yes, I have the exact same problem. :( -Original Message- From: Tomislav Vojvodic [mailto:[EMAIL PROTECTED] Sent: Thursday, May 11, 2006 5:48 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted - omniis.com. Version is 0.10 (newest) Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup' isn't even implemented in AstTAPI driver so that could be the reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T.S Sent: Thursday, May 11, 2006 1:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial I had similar problems when I first started to play with it. I've gotten Omniis TSP for Astrisk to work just fine. http://www.omniis.com/asttapi But i don't know the version im using 0.0.8 Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Wednesday, May 10, 2006 2:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hello, I'm experiencing some problems with AstTAPI driver. Dialing works just fine, but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact that Outlook doesen't detect end of conversation - once the call is terminated 'manually' via the phone Outlook still 'thinks' that call is active. Anyone knows what's the problem? Is 'hangup' implemented in AstTAPI driver? Thanks, Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1530 (20060510) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1532 (20060511) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Asttapi for Asterisk 1.2 Testers Needed (was RE:[Asterisk-Users]Asterisk TAPI - Outlook click2dial)
Had I have been smart originally I would have done this to start. Some rudimentary documentation above and beyond Asttapi 0.10's poor documentation is available along with the download at http://www.kirkhamsystems.com/asttapi. Clint -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Tuesday, May 16, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: Asttapi for Asterisk 1.2 Testers Needed (was RE:[Asterisk-Users]Asterisk TAPI - Outlook click2dial) Please hook me up. I have customers dieing for something that works. All our systems are Asterisk 1.2.7.1 -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clint Sharp Sent: Tuesday, May 16, 2006 11:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Asttapi for Asterisk 1.2 Testers Needed (was RE: [Asterisk-Users]Asterisk TAPI - Outlook click2dial) I've finished a patched version of asttapi that will work with asterisk 1.2. There were fundamental changes to the Asterisk Management interface between 1.0 and 1.2 that broke asttapi. I think my patched version will work on 1.0 and 1.2 branches, but I have no way of testing since I don't have a 1.0 install nor do I want one :). I'm looking for testers, if anyone's willing to test this out, I'll send you a zipped copy of the TSP file (I haven't worked on doing an installer yet). I need to send out the debug build so I can generate information in case it doesn't work on anyone else's PC. Contact me off-list for copies. Clint -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Friday, May 12, 2006 1:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Oh.. :/ too bad.. I'll have to look at the source.. bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T. Shaw Sent: Thursday, May 11, 2006 11:20 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Yes, I have the exact same problem. :( -Original Message- From: Tomislav Vojvodic [mailto:[EMAIL PROTECTED] Sent: Thursday, May 11, 2006 5:48 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hey, thanks for your reply.. ;) I'm also using asttapi from website you posted - omniis.com. Version is 0.10 (newest) Well yeah.. the problem is that hangup doesen't work. Maybe 'hangup' isn't even implemented in AstTAPI driver so that could be the reason why Outlook+AstTapi doesen't know what 'Hangup' from Outlook is. When I clik 'Hangup' in Outlook there is nothing in Asterisk debug/cli window. Only problem is that Outlook still thinks that call is active even if you hangup the phone manually.. I mean, when I put the earphone back to base/station/phone.. whatever. Dialing works just fine. Because of that you need to close that window 2 or 3 times if you want to call same person/contact again. Bye, Tomislav -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of T.S Sent: Thursday, May 11, 2006 1:08 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk TAPI - Outlook click2dial I had similar problems when I first started to play with it. I've gotten Omniis TSP for Astrisk to work just fine. http://www.omniis.com/asttapi But i don't know the version im using 0.0.8 Terrelle -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomislav Vojvodic Sent: Wednesday, May 10, 2006 2:23 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk TAPI - Outlook click2dial Hello, I'm experiencing some problems with AstTAPI driver. Dialing works just fine, but 'Hangup' from Outlook doesen't.. actually that's not the problem as fact that Outlook doesen't detect end of conversation - once the call is terminated 'manually' via the phone Outlook still 'thinks' that call is active. Anyone knows what's the problem? Is 'hangup' implemented in AstTAPI driver? Thanks, Tomislav ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http
Re: [Asterisk-Users] GXP-2000 Volume Issue
I sent this from the wrong address and I don't think it went through. I've just done some testing on the phone on 1.0.1.9 and 1.0.2.13. The one on 1.0.1.9 has no outbound gain issues, it is nominal with the rest of the phones in out office (Snom 320, Polcyom IP 301, and Budgetone 101). However, this one on 1.0.2.13 caps at about a third of the meter on ztmonitor. Is anyone else having this issue, or might this be a hardware issue with this particular phone? ClintOn 3/1/06, Clint Sharp [EMAIL PROTECTED] wrote: I have one on 1.0.2.13 and one on 1.0.1.9. The one on 1.0.2.13 is the one I can imperically say is too quiet, the other appears to be better. I went back to 1.0.1.9 on the other because of a handset volume issue. ClintOn 3/1/06, Paul C [EMAIL PROTECTED] wrote: I had the opposite problem, I had to set txgain down as they were too loud and causing problems. - Original Message - From: Clint Sharp To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, March 02, 2006 6:56 AM Subject: [Asterisk-Users] GXP-2000 Volume Issue Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the Asterisk side, or does anyone know of any gain adjustments that can be made to the GXP-2000s on either the older 1.0.1 series firmwares or the new 1.0.2 branches?Clint ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP-2000 Volume Issue
Is anyone else having an issue with GXP-2000s and transmit gain? All my other phones are fine on my TDM400P with txgain set at 0, but the GXP-2000 caps at about a third of the scale in ztmonitor. I'm getting people complaining they can't hear me on my GXP-2000s, whereas my Snom 320 and Polycom 301 are great, and my Budgetones are overmodulating. Is there any conceivable fix on the Asterisk side, or does anyone know of any gain adjustments that can be made to the GXP-2000s on either the older 1.0.1 series firmwares or the new 1.0.2 branches?Clint ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
I had to drop 1.0.1.12 because it has a serious handset volume issue that seems to cut the handset volume in half. Fix one bug, cause another. Clint On 2/22/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Wed, 22 Feb 2006, Clint Sharp wrote: 2) GXP-2000: Not much better than the Budgetones, but at least the firmware [...[ that phone's quality).The speakerphone is useless due to echo issues. speakerphone echo bug was fixed in 1.0.1.12-Dan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on SPA-3000 rings extensions only after third ring
I also have the same issue on a TDM400B. This isn't a valid solution for us, because I don't want to give any sort of large business feel to our system (any sort of please hold kinda stuff would really turn off customers who are used to just getting an answer, even if it starts ringing again). I've set the start parameter in zapata.conf down to 300 from the default of 1500, and that seemed to have helped, but it's hard to tell. Anyone else have any different solutions to this? Do I have to turn off callerid to fix this problem?Clint On 2/21/06, Joseph Tanner [EMAIL PROTECTED] wrote: Same problem here with X101P.In my case (and I'm guessing yoursituation is similar, but not sure since the hardware is different)asterisk needs to see the first ring come through before doinganything.Sometime between the first and second ring it gets callerid information, then sometime after the second ring it can startprocessing the call (i.e., execute what you have set inextensions.conf).My workaround, is to have asterisk actually answer the call, and a voice (which my lovely wife recorded) tells the caller to please holdwhile their call is connected.Once the caller hears ringing tonesagain, your internal lines are ringing at the same time.No morehaving people hang up after 4-5 rings, when all you've heard is 1-2. This setup has the nice side effect of letting me force unknowncallers to press 1 before being connected.Anyone I know (and haveentered their phone number in extensions.conf to recognize) won't haveto dial 1.All others will, and this has so far eliminated all telemarketing calls and even all wrong numbers (they know right awaythey got the wrong number, and hang up without pressing 1).It also lets me gain access to various functions no matter where I'mcalling from.I can enter a password while the recording is playing, and get dialtone.From there I can call out (like a calling card), orcheck voicemail, etc.It's just like I'm dialing from an internalextension, which can come in handy (say I need to reach my wife in the middle of the day, when she's usually asleep and GotoIfTime directsthe calls to voicemail, and I'm calling from an unknown number; I justenter my password, get dialtone, dial 6 which I have setup to ring all internal extensions regardless of time, and voila! I have a grumpywife).If asterisk didn't automatically answer the call, none of thiswould be possible (well, I suppose I could press 0 during voicemail, and have the o extension setup, but this way works better).Maybe not the solution you were looking for, but personally I thinkthis workaround opens up a lot of possibilities you may not havethought of previously. Joseph Tanner ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What business IP phone to use
It's funny this thread has been coming up, because I've been testing out phones at my office, and I just did a fairly intensive quality test on them.1) Budgetones: Don't bother for a business setting. The speaker phone is basically useless (echo problems) and the handset is horrible. If you follow the suggestion on the Wiki to drill out the handset, it improves things marginally, but not much. Users talking to you will constantly complain about you sound muffled. It's think it's a frequency response thing and not a volume thing, I think it's just getting lower than a standard 8 khz sample out of the microphone, because it's so cheap. 2) GXP-2000: Not much better than the Budgetones, but at least the firmware is still in active development. Feature-wise it's pretty cool, but poor firmware and poor handset hardware again make this a real problem for us. We lost one handset to static electricity yesterday (which was fixed by adding in a microphone from an old business set, which actually improved that phone's quality). The speakerphone is useless due to echo issues. However, 4 line appearances is pretty cool for that price of phone, and passthrough Ethernet at 100 mbs is pretty cool too. Overall, I can't recommend them, because while they sound slightly better than the budgetones, I still get many complaints about muffled calls. 3) Polycom: Of the 4 phone brands we're actively using (not including the Wifi phone which rarely gets used), this was the best until I got the Snom in today. The handset is of good quality. I have an IP 301, but if the cheapest phone is this good, I'd definitely get a 501 or 601 (and am considering ordering some, although I may order Snom 320s instead). Their support policies do get on my nerves, I'd like to not have to worry about what reseller I'm using, but it's a solid phone with solid features, although the menus are cumbersome and I haven't gotten MWI to work on it yet. 4) Snom 320: This is an excellent phone based off one days testing. Minimal configuration, professional looking web interface, and the best sound quality of any of the phones I tested. THe speakerphone works great, and the handset quality is outstanding, and tested the best with my callers that were listening to me through the PSTN. I haven't upgraded firmware or anything on this yet, so can't tell you there, but I can't see a compelling reason to upgrade from whatever it shipped with that this point (i'm not feature crazy, I only upgrade the firmware if basic features don't seem to be working right). Overall, stay away from the Grandstream's IMHO. The audio quality issues will drive you insane. I'm hoping someone will come out with a sub-$100 phone that drops some features but fixes what should be the cheapest part of the phone to manufacture, since they've been the same for nearly 50 years, the handset. Clint ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
If you can live without a speakerphone, the Polycom 301 is an excellent phone and is only $30US more expensive than the GXP-2000s. I tend to trust the Polycom brand, and they haven't really steered me wrong yet in the IP phone hardware. I'm interested though in any reports of success with the Aastra phones, as I'm certainly looking for something with better quality than Grandstream with lower-end pricing. Anybody have any reports on using the Uniden phones? ClintOn 2/18/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: why bother with an aastra 9133i when you can have a polycom 501. betterphone, same price. -Dan___--Bandwidth and Colocation provided by Easynews.com -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bridged line appearance
I'm having a very hard time justifying trying to sell this to the SOHO market on price or parity with key systems. I've installed key systems and large scale PBXs, and while working around the SLA problem isn't that hard, the price point for a key system is very hard to compete with. I've never understood why people would want to use an SLA system, honestly, as it's a really poor model. I hate sitting in offices with constant paging Call for blah, line 1. The PBX model to me is much more preferable, and working around it is simply a training problem. The problem with asterisk isn't the lack of SLA, it's the price point. It's going to be very hard for IP phone vendors to compete on price at this point, and so far the quality issues in low-priced hardware to me means I can't really sell this to anyone who's not willing to pay $200-$300 a phone (retail). Not that it's impossible, it's a different sales strategy. Perhaps people who are wanting to sell this to the SOHO market should attempt to change the game, as PBX like functionality doesn't exist in the SOHO market because it hasn't been affordable previously. Asterisk systems are pretty cheap in terms of the features they offer, such that the sales pitch really depends on cost for features and maintenance and infrastructure savings rather than overall cost. Admittedly though, the voicemail system's navigation issues are a big problem. ClintOn 2/18/06, John Novack [EMAIL PROTECTED] wrote: Many very low cost hybrid key/pbx systems for the small business SOHOmarket have 12 or more programmable buttons, so regardless of what isdone with Asterisk, until the IP phone manufacturers take off their blinders and manufacture competing equipment, this market will be out ofreach. These same systems now have voice mail systems with capabilitiesand features that make Comedian Mail the correct name.Asterisk isn't alone regarding these shortfalls, of course. IP phone system designershave failed to understand the small business market for several years.Defensive responses with lengthy explanationswhy it can't be done, or why it hasn't been done and will be difficult miss the point. EitherAsterisk needs to change to move into this market, or another product willJMOJohn Novack___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream GXP-2000
I'm still having numerous echo issues, even on SIP calls, with the GXP-2000s. Unfortunately, they cause echo on the remote end on SIP calls, which does not occur on other phone models. The speaker phone is unusable due to echo problems. Maybe the 1.0.2 firmware branches will help, but I'm scared of upgrading with no path back to a stable firmware. They're really nice hardware, but unfortunately the software for them just stinks (no gain control on the handsets or speakerphones, lots of missing options I'd like). Unfortunately, I have yet to find a sub $100US phone that I like. I definitely would not order 15 until I'd ordered a couple and tested. ClintOn 2/17/06, Mimmus [EMAIL PROTECTED] wrote: Hi,I'm going to propose to my boss the buying 15 Grandstream GXP-2000 phones.- Is it a good choice (budget limit of 100 Euro/phone is mandatory)?- Can be a profitable business the direct buying of 50 phones (to save other money) or is it a risk?Thanks in advance--Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] more cpu intensive echo cancellers ?
Gerard Saraber wrote: Thanks! testing it now, on my test calls it appears to start out with less echo then the Mark3 canceler, but it trains slower, seems like it took a long time for the echo to completely disappear, the real test will be seeing what the people at my company have to say. Feb 9 14:47:51 [kernel] Zapata Telephony Interface Registered on major 196 Feb 9 14:47:51 [kernel] Zaptel Version: SVN-trunk-r934M Echo Canceller: MG2 I've had really good luck with the echocan preload patch that was posted on the asterisk dev list a while back as well, and I've been recommending it to people as well. This has really helped minimize the echo problems to a minimal level, although I don't know about recommending this system to our customers. I still think a lot of my audio quality problems are being caused by my phones (not echo, but clicks and pops and various overmodulation problems). We're getting there, but I'm still nervous with trying to sell an * system to someone who is used to the quality of a traditional PBX or key system. Clint ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agressive echo cancelation
Ronald Hartmann wrote: Anyone know if it is possible to control how aggressively the Aggressive mode behaves. Meaning, is it possible to dial back the aggressive mode to have a happy medium between Regular and the Aggressive defaults. I have a situation where Normal echo cancellation is not quite enough, however when I turn on aggressive mode We are attacking it to hard and I am unhappy with the walkie talkie behaviour of the Aggressive mode. Thanks for any feedback. ~ron Not that I could see in the headers, I didn't check out the source code. I loaded the recent Echocan Preload patch up and with MG2 echocan, the preload setup after dumping a call that had time to train well (about 30 seconds) and loading those values, and then having the gains set down pretty low I'm able to get decent quality. The final problem I fixed on Friday was putting 1.0.1.9 back on my GXP-2000s since 1.0.1.12 was way too quiet on the handset and turning the gain values up really horked up the echocan. Clint ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
