Re: [Asterisk-Users] BAD/GOOD Echo Cancel

Rich Adamson wrote: More than that, in their fine print some only claim to pass maybe two or three of the tests. There is nothing that defines what you must achieve before you can claim G.168-2002 compliance. Well, isn't that just wonderful :-) Standards are amazing things, from a

Re: [Asterisk-Users] BAD/GOOD Echo Cancel

For the record, I've done a couple of Asterisk installs, and HATED echo -- or feebly attempting to get Asterisk's flakey software algorithms to do anything about it. Finally got sick 'n tired, and threw money at it -- got the Sangoma quad-span T1 card. And echo freaking VANISHED. (Note that,

[Asterisk-Users] MWI on Polycom 501.

Hi! I've got MWI working just fine for my 501, but it's on if I have -any- VM messages. I only want it on if there are *new* messages. Any ideas as to what I should be changing? Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] MWI on Polycom 501.

Anthony Rodgers wrote: Hi Ken, When you say -any-, what do you mean? Messages in the Old folder, or what? Precisely. If there are messages in the Old folder, the MWI still blinks. (I suppose I should've been more explicit; apologies...) -Ken ___

Re: [Asterisk-Users] ZAP -- sip(polycom301) can not hear each other

From your description, it sounds as if the SIP phones are local to the Asterisk box. If this is so, having nat=yes might be a problem. -Ken sdgesa gaeharth wrote: please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the

[Asterisk-Users] Forwarding issue.

If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI -- SIP read from

Re: [Asterisk-Users] Polycom IP501 Endless Loop

I had the /exact/ same problem. Turns out it's the FTP server; in the docs, there are several FTP servers specified as being compatible; proftp is the one I went with, and it fixed it right up. (Note that I was using the default Debian FTP server when it was rebooting, so it's not just a 'doze

Re: [Asterisk-Users] Forwarding issue.

-- as the .sample file suggests -- appears not to work for Polycoms.] -Ken Regards, - Brad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Tuesday, January 31, 2006 11:20 AM To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject

[Asterisk-Users] DID over analog?

I've some DID's that I'm using for in-bound faxing, but I'm having some trouble with getting that working perfectly on my T1. So I'm thinking of pointing them to an analog line. Will the DID's simply come in over the analog, presumably sending the DID digits via DTMF? Or is that not something

[Asterisk-Users] Dynamically disabling echo cancellation (Zap).

Hi! For reasons that I won't bore people with, I'd like to disable echo cancellation on-the-fly, depending on which DID a call came in on. I've seen things like spandsp disable EC for faxes, so I know it's possible. Any idea where to start looking? (I assume I'll have to make a helper

[Asterisk-Users] Received fax offset in tif file?

Hi, all -- some (but not all) of the faxes I receive on my Sangoma card have an offset, where the right 3/4 of the fax are shifted left, and the remaining portion is pasted on to the right edge; see jots.org/~ken/fax.jpg for an example. Anyone have any ideas on how to get this working? Since I'm

[Asterisk-Users] Re: Polycom boot times/XML files.

Andrew Furey wrote: Huh? My 7905 takes well under 10 seconds, including Asterisk registration and NTP update. Granted, if it were DHCP it might take marginally longer, but 5 _minutes_? Yeah, the Polycoms *do* take a while to boot -- but not five minutes. I've timed mine (Polycom 501's) and

[Asterisk-Users] Extensions for in-bound faxes w/o properly-provisioned T1.

Hey, all. I've got a non-PRI T1 that doesn't do DID correctly: I can't get the DID from the proper variables, and, instead, I direct it based on the four least valuable DTMF digits dialed by the T1 for in-bound calls. Which really works pretty well; Asterisk plugs them quite nicely into

[Asterisk-Users] SpanDSP not sending to fax extension.

. Everything seems fine, except that it never gets recognized as a fax. I've even turned off echo cancellation -- same deal. Any ideas? I'm pretty stumped, here. Thanks! -Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com

Re: [Asterisk-Users] SpanDSP not sending to fax extension.

, Ken D'Ambrosio [EMAIL PROTECTED] wrote: Hi, all. I've got a fax extension in my extensions.conf, but spandsp never sends my faxes there. Both applications -- txfax and rxfax -- are registered by Asterisk, so they compiled and installed correctly. I've got a Sangoma A104 card, and (as some

[Asterisk-Users] MeetMe greeting message.

Hi, all. My president wants to have a custom greeting for our bridge. So, I had it recorded (as foo.gsm), modified app_meetme.c to reflect the new filename, compiled, installed... and now get Jan 16 12:53:05 WARNING[14859] file.c: File foo does not exist in any format Jan 16 12:53:05

[Asterisk-Users] Decent sub-$100 SIP phone.

anyone have any suggestions for decent sub-$100, professional-looking SIP phones? Thanks! Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Immediate routing on 0 (DNIS)?

I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the range of 0600 - 1699. However, the second that the 0 is seen on an in-bound 06xx call, it stops listening for any more digits, and immediately tries to route the call. My 16xx numbers wait for all four digits before trying

Re: [Asterisk-Users] Immediate routing on 0 (DNIS)?

suggestions/ideas/etc... -Ken Ken D'Ambrosio wrote: I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the range of 0600 - 1699. However, the second that the 0 is seen on an in-bound 06xx call, it stops listening for any more digits, and immediately tries to route the call. My

Re: [Asterisk-Users] Immediate routing on 0 (DNIS)?

until it had all the DID stuff. I guess DTMF is DTMF, regardless of the source, huh? In which case, I'm all set -- again, thanks for setting me on the right road! -Ken Thanks for any suggestions/ideas/etc... -Ken Ken D'Ambrosio wrote: I've got a T1 (EM wink). Our four-digit inbound

[Asterisk-Users] Polycom 501 netboot not working.

When I try to boot my 501, it runs through the usual stuff, then stops with Config file error Error is 0x4020 and then reboots. The log on the FTP server shows: 0105164151|app1 |3|00|Bootline: ircaIP 0105164155|cfg |3|00|Image bootrom.ld has not changed. 0105164159|cfg |3|00|0004f202f803.cfg

Re: [Asterisk-Users] Polycom 501 netboot not working.

there. I've reset the phone with 4-6-8-* keys, but same thing. I'm tempted to try another phone, and see if I get anywhere. But before I -kill- another phone, I thought I'd ask if anyone else has seen this or anything like it... -Ken On Jan 5, 2006, at 1:37 PM, Ken D'Ambrosio wrote: When I try

[Asterisk-Users] Screening incoming calls.

The PBX I'm getting ready to replace has a really nifty feature -- one that I'm not even sure Asterisk -could- do -- though I'm hoping to be proven wrong. When a call goes to voicemail, the end-user can listen to the VM as it's being recorded, and can interrupt and answer the call if it's someone

[Asterisk-Users] Screening incoming calls.

The PBX I'm getting ready to replace has a really nifty feature -- one that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven wrong. When a call goes to voicemail, the end-user can listen to the VM as it's being recorded, and can interrupt and answer the call if it's someone

[Asterisk-Users] DNIS dropping digits.

I've got an EM wink T1 into a Sangoma card, and it's dropping my DNIS digits. I'm supposed to get four digits from the CO, and I reliably get the first digit -- it's a crapshoot as to how many after the first I get. I have overlapdial=yes and immediate=no. Any other suggestions of things to

[Asterisk-Users] Which Asterisk GUI?

decent, but there are a lot of things it doesn't do, too. So: which GUI do -you- like? Thanks! -Ken D'Ambrosio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

Re: [Asterisk-Users] GXP-2000 any good with * ?

Anyone using the GXP-2000 with * ? Yup. I like the phone -- works nicely, easy to configure, four lines, etc. Any showstopper problems? Well... see below. The echo issues, is it speakerphone only? The speakerphone kinda sucks. Maybe someone with a newer firmware or different config

[Asterisk-Users] AMP stuff via CLI?

! -Ken D'Ambrosio ___ --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] Can Asterisk act as a media gateway?

I've got an account that's looking at doing some cable/VoIP integration. They were wondering if it were possible to set up something like this: PSTN (T1) - Asterisk - (some VoIP protocol, probably SIP) - Siemens soft switch - their product It sure sounds nice in theory, but I've never

[Asterisk-Users] DSP-based echo cancellation (T1).

to. Suggestions? Thanks! -Ken D'Ambrosio ___ --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] Two POTS in, but only want one out?

Hi! I've got two POTS lines coming in to my * box, but I only want the primary of the two lines available for outbound dialing. I can't quite figure out how to make that happen. Suggestions? Thanks, -Ken D'Ambrosio ___ --Bandwidth and Colocation

[Asterisk-Users] Not answering/script.

where to start with something like this, but I have to imagine it's been done before. Thanks! Ken D'Ambrosio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] Asterisk Vs. Cisco, et. al.?

I know the reasons I like Asterisk -- and that's all well and good. But if I'm bidding against (say) Cisco, money aside, what should my argue points be? Is there a compare-and-contrast Asterisk-vs-the-other-guys page out there somewhere? Just curious, -Ken

[Asterisk-Users] Dialing w/analog phone via FXS port.

Argh. I can't figure out what I'm doing wrong. I can dial with my SIP phones just fine, but I want to set up an analog phone plugged into my FXS port... and, while it gets dialtone, no matter what digit I press, I get stuff like: VERBOSE[21963]: -- Starting simple switch on 'Zap/1-1'

[Asterisk-Users] Calls from analog/FXS phone?

I've got a 400P, with a couple FXO's and and FXS. I've got an analog plugged into the FXS, and it gets dialtone fine. However, whenever I press any digits, I get the doo-dah, doo-dah unhappy sound. I've got a functioning FXS system at home, but I was trying to plug this into an AMP-created

[Asterisk-Users] Reproducible echo on IAX calls to -some- destinations.

I'm very, very confused. Dialing out, through VoicePulse, with both gsm and ulaw CODECs, most of my calls are great. However, calling my (non-Asterisk) voicemail at my job, and calling my cell phone both produce horrendous (~ 1/3-second delay) echo. I've tried with different phones (Polycom

Re: [Asterisk-Users] Polycom Soundpoint boot ROM upgrade: how?

Put the new bootrom.ld and bootrom.ver files on the config server (FTP or TFTP) that your phones load from, and they will upgrade automatically. Easy. Too easy. ;-) Seriously, though: I'd've never thought of that. Thanks much! -Ken ___

[Asterisk-Users] Polycom dhcpd.conf? [Or, Some day, I'll figure this all out.]

I'm RTFM'ing, but I can't figure out how the dhcpd.conf file specifies the boot server, and how it differentiates between whether it's FTP or TFTP. I've tried option 66/next-server, and option 150, to no avail. And the docs just don't -- leastwise, in the way I'm reading them -- make sense.

Re: [Asterisk-Users] Polycom dhcpd.conf? [Or, Some day, I'll figure this all out.]

Interesting... I'll stick with FTP anyway, since I can partially secure it, and it works across NAT :-) Kevin: I don't think Matt made himself quite clear enough -- going with his dhcpd.conf setup works for FTP, regardless of what the option name is; it's working great! And I don't even have

[Asterisk-Users] Polycom Soundpoint boot ROM upgrade: how?

I can't figure out how one upgrades the boot ROMs for the Polycoms. I've read the wiki, I've checked the docs, I've looked here, there, everywhere; I hoped it might be through the web interface, through DHCP... but I can't see how to do it. Suggestions? Thanks! -Ken

[Asterisk-Users] XML config files for Polycom SoundPoint IP 300?

I bought a couple Polycom Soundpoint 300's, and have them working nicely with SIP... but I'd like to be able to do automatic config via FTP, but it requires some XML config files. The docs discuss them in detail, but I can't seem to d/l them from Polycom. [No, it doesn't appear to be on the CD

[Asterisk-Users] No compatible codecs! -- worked with 1.0.0, not 1.0.6 or CVS.

files for both; I've tried enabling/disabling ULAW, ALAW and G279 on the Soundpoint, to no avail. Plug 1.0.0 back in -- works like a champ. To say that I'm confused would be understating things rather severely. Thanks much, Ken D'Ambrosio ___ Asterisk

Re: [Asterisk-Users] No compatible codecs! -- worked with 1.0.0, not 1.0.6 or CVS.

To say that I'm confused would be understating things rather severely. Kevin P. Fleming wrote: To say that we can't help you without seeing your config files would also be an understatement. Unfortunately, we are not all-knowing nor telepathic, so just saying it doesn't work won't generate

[Asterisk-Users] Re: Linux Bridge + QoS Shaper HOWTO available

Howdy! I'm VERY interested in your HOWTO... but the link you have, below, times out. Any chance you could mail me the HOWTO, or point me to a new link? Thanks much! -Ken [EMAIL PROTECTED] wrote: I've created a pretty complete HOWTO on creating a Linux Bridge (using Fedora) to shape LAN --

[Asterisk-Users] Polycom 300 -- No compatible codecs!

I've got all three CODECs the 300 supports -- G.711u, G.177A, and G.729AB -- enabled, I've changed the order, I've got them all in allow lines in my sip.conf, as follows: disallow=all allow=ulaw allow=alaw allow=G729 From sip debug I get the following snippets:

[Asterisk-Users] Echo on SIP -- not on analog.

Okay, I'm stumped. When I call the PSTN (through POTS lines), my analog phone phone works fine. My SIP phones -- a Grandstream and a Polycom -- have major echo; roughly a .25 second delay. Eventually, it goes away, which I guess is echo cancellation in action. But, dammit, why does my

[Asterisk-Users] Asterisk on a notebook?

I'd dearly love to be able to give an Asterisk demo by just toting my notebook, a PC/PCMCIA card, and a couple SIP phones. Is there any way to do this? Or should I look for a small-profile box with PCI slots, instead? ___ Asterisk-Users mailing list

Re: [Asterisk-Users] DHCP Attribute for TFTP server for Aastra 480i?

Olson, Dana wrote: I have this Aastra 480i phone, and you can set the TFTP server IP address manually in the phone, but there should be a way to have it find the TFTP server information via DHCP. Does anyone know if this is possible, and if so, what is the attribute I have to set on my DHCP

[Asterisk-Users] Sipura 841 delayed: other PoE options?

I -think- that the Sipura 841 was PoE... and I'd been anxious to find out. However, according to Atacomm.com, it's been delayed until mid-January. *sigh* So: does anyone know of a (decent) phone that meets the following criteria, and isn't too expensive? - SIP - two (or more) lines - some form

[Asterisk-Users] Should echo cancellation be a science or an art?

Perhaps 90% of my calls -- over a Uniden and a Grandstream -- are fine. The other 10% get some nasty echoes. Is there some magic something I should be tweaking? I kind of thought that echo cancellation was static, rather than dynamic, and that it's difficult to be able to cope with something

[Asterisk-Users] Comdial PBX -- can use Asterisk as VM box?

Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a KeyVoice application -- is dying.

Re: [Asterisk-Users] Problem with Grandstream bt100

R A wrote: Then what do you think i have to do? i install a sniffer and the phone make an ARP request to 67.153.142.69. the phone is 192.168.0.160 and i set my pc to 192.168.0.161 and i can't ping the phone. Get a new phone. :( What he's saying -- and I agree with him -- is that, if you

Re: [Asterisk-Users] Uniden UIP200 -- configured, but not working?

Ryan Courtnage wrote: If it registers fine with Asterisk, then what exactly is the problem? Does the Uniden phone display an error? Asterisk? Can you make/receive calls? The firmware version and unidenmac.txt might also be relevant to the problem. Jeepers. You want a description of the

[Asterisk-Users] Uniden UIP200 -- configured, but not working?

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden

[Asterisk-Users] T100P -- data?

, that's readily findable). Thanks much, Ken D'Ambrosio ___ 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] Uniden UIP200 configuration -- manual MIA?

, maybe I'm just dumb, but I'm thinking that there's an admin manual I didn't get. Anyone have any pointers? Either web-wise, or simple stuff like the password for the unlock config option? Thanks much... Ken D'Ambrosio ___ Asterisk-Users mailing list

[Asterisk-Users] Digits being lost going out POTS line?

When I dial out, from both my analog and my SIP phones, it fails roughly 80% of the time with various telco messages (eg., The number you have dialed...). The messages take a good 30 seconds before I hear them, which makes me think the telco isn't seeing enough digits. 20% of the time, my calls

[Asterisk-Users] Asterisk on a mid-sized flat corporate network?

! -Ken D'Ambrosio ___ 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] SIP outbound dialing -- newbie alert.

(and maybe even inbound ;-) calls, so I can get something working, and build from there. Any suggestions? If it means I need to go over docs I've already read, that's fine, but I'm pretty confused right now... Thanks, - Ken D'Ambrosio ___ Asterisk-Users

[Asterisk-Users] Three (quick?) questions...

I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users

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