RE: [Asterisk-Users] Manager API Call Origination Variables
I'm using this in production, and it works like a charm. (example from PHP). Phone numbers have been changed to protect the innocent. ;) fputs($socket, Action: Originate\r\n); fputs($socket, Channel: Zap/g1d/1234567890\r\n); fputs($socket, Exten: 5002\r\n); fputs($socket, Priority: 1\r\n); fputs($socket, Context: userexten\r\n); fputs($socket, Variable: dialnumber=76|confnumber=3236\r\n); fputs($socket, CallerID: Dev Meeting 760-000-\r\n\r\n); Brian D'Arcy Operations Engineer Akiva Corporation -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Peter Osborne Sent: Monday, November 15, 2004 2:09 PM To: [EMAIL PROTECTED] Cc: Peter Svensson Subject: Re: [Asterisk-Users] Manager API Call Origination Variables Well I tried just about every combination that I can think of as well as every combination mentioned and it still doesn't work. Not sure why, maybe it's just not possible from the Manager API. Pete On Monday 15 November 2004 04:56, Peter Svensson wrote: On Mon, 15 Nov 2004, Brian West wrote: Ok to cut confusion here Its: Variable: _ALERT_INFO Value: somevalue Its always var/val via manager. Not in the Originate action it isn't. This is what both the help show manager command originate say and what reading the source indicates. From the help: Variable: Channel variable to set (VAR1=value1|VAR2=value2) Peter The channel does not exist prior to the Originate action. However, you may be able to pass variables in the originate command itself: Action: Originate Channel: sip/12345 Exten: 1234 Context: default Variable: _ALERT_INFO=Ring Answer|SomeOtherVar=SomeOtherValue This may work. Peter Peter ___ 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 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 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
RE: [Asterisk-Users] Debian Sarge -- cvs vs. apt-get
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Roberts Sent: Thursday, September 09, 2004 2:30 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Debian Sarge -- cvs vs. apt-get After reading the wiki, I am still confused as to whether I should install zaptel, libpri, and asterisk from cvs or use apt-get to retrieve the Debian packages. I am running Debian sarge with kernel 2.4.26. What is the prevailing opinion out there about using cvs vs. apt-get for the various components (zaptel, libpri, and Asterisk)? Thanks, Mike Roberts Mike, I use Debian with the same kernel, and run Asterisk in a corporate production environment with it. I would suggest using CVS. It gives you a much greater control over your deployment. It took a little work on debian getting all the appropriate packages installed, but IMO, it was worth it. Brian D'Arcy ___ 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
RE: [Asterisk-Users] PRI issue
Ben, I ran into a similar issue on the 8/31 cvs, except it was backwards. Outbound calls would report a busy on the channel selected, yet a few minutes later the channel would be used for an inbound call. I had to revert back to my previous checkout from 8/16 to resolve the issue. The problems didn't break the channels completely, it happened probably every 5-10 minutes. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Wednesday, September 08, 2004 7:51 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI issue Hi, I recompiled asterisk today from CVS and I've been having a number of problems, I've read the deadlock page on the wiki and some of it sounds like that, however, the latest issue we're having it that sometimes Asterisk doesn't seem to know the PRI channel has dropped, and assumes it's still busy. However, that same channel can be used to make an outgoing call?! Has anyone experienced anything similar? Regards, Ben ___ 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
RE: [Asterisk-Users] Cisco 79xx series IP phones
Shawn, That's a complete load of manure. I have an office full of 7960's, they work great with asterisk with the SIP images loaded. I'm about to pick up a lot of 7912's (simple one line phones, same as the 7905 but it has a built in switch). These phones have also been confirmed to work with Asterisk. I would recommend not going directly to cisco, and just find a reseller who can offer you the phones preloaded with SIP images, or the phones + the smartnet contracts so you can get the images and firmware from cisco yourself! I've tried a *lot* of phones with Asterisk, and thus far, the Cisco's are by far the best I've used. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shawn Parker Sent: Friday, August 13, 2004 9:31 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 79xx series IP phones I got a call from our Cisco rep today saying that they couldn't sell just phones to anyone because if my ethernet isn't to exact spec... then they won't work at all. I've read over the Wiki documentation and it seems that the 79xx series phones work with Asterisk. They told me that without a Cisco phone system in place or a Cisco router or switch, then the ethernet wouldn't work with the phones. Is this true, or is it someone just trying to sell me a Cisco system? I don't see how my use of a Planet or Netgear switch would alter the spec of my ethernet to cause a IP phone to fail. Seems far fetched to me. I've never had any other problems mixing Cisco equipment with other product lines. Does anyone have any knowledge or experience to give me dealing with Cisco 7902G and 7905G IP phones and getting them to work on a lan with Asterisk when *not* using other Cisco hardware? Cheers, Shawn ___ 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 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
RE: [Asterisk-Users] Rodopi Billing
On Fri, 30 Jul 2004, Darren Bentley wrote: Hello, Has anyone used Asterisk in conjunction with a billing system like Rodopi? Is the Rodopi VOIP module worth getting, or can radius be used? I suffered with Rodopi for three years in a previous life. Avoid it like the plague. OMG.. I had to support a rodopi installation myself for 2 years.. Closest I've ever come to suicide. While I have not managed another system but RODOPI, I have to say, there must be better. ___ 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
RE: [Asterisk-Users] Best Linux for Asterisk
Hi Andy, Before Asterisk came into my life, I hadn't used Linux since RedHat 4.7. I did some research and decided to use one of the debian netinst images this go around, and I couldn't be happier. While it took me a day stumbling thru the packages and re-learning my way around, figuring out dependencies to get everything compiled and working etc... I've gotta say that the Asterisk + libpri + zaptel + tts stuff is rock solid, as is the system. I'm running Debian Woody with the 2.4 kernel. This system is also in a heavily used production environment within a software company. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Kirkland Sent: Wednesday, July 28, 2004 6:14 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Best Linux for Asterisk Hi folks; Can anyone recommend the best Linux OS (versions, etc) to run Asterisk? I'd like to be able to run the Text To Speech apps and some of the extended functions of the software (no phone hardware needed, all Voice over IP stuff)... I'm currently running Asterisk on Mandrake Linux (vesion 10 I think?) but I'm having difficulty compiling the TTS stuff. I'm just wondering if there's a widely used version that pretty much works with everything...? Andy --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004 ___ 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 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
RE: [Asterisk-Users] Open for beta testers - free calls in us/canada
Hello Sean, Thanks for the opportunity to test your service, along with the free long distance! What codecs does your system support currently? I am setup to allow g711u, g729 and GSM, however I'm seeing a lot of codec errors when connecting through your service (using SIP) Thanks Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kane Sent: Tuesday, July 27, 2004 10:28 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Open for beta testers - free calls in us/canada We have another 500 beta openings in the SimpleConnect beta. SimpleConnect is a service for you to make IAX/SIP calls from * or any IAX/SIP agent. Beta participants get free calls to anywhere in the United States and Canada. If you want to become a beta tester, just go to https://secure.simpletelecom.com/order/ . No credit card is required. We're looking forward to your feedback. Sean SimpleConnect ___ 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 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] Strange behaviour using 7960
Hello all, One of my remote employees is using a 7960 we sent him, on a public IP address at his home office. I've run pings and traceroutes both from the server to his phone, and from the cable modem to our server, there's never a high ping time, or a dropped packet, however about every 30 minutes to an hour into his calls (not all of them, it's random) he can no longer hear the opposite party, but they can hear him.. The messages in the asterisk console always look like this: Jul 20 12:41:34 WARNING[14350]: Maximum retries exceeded on call [EMAIL PROTECTED] Jul 20 12:41:34 NOTICE[14350]: Peer 'dnicol' is now UNREACHABLE! Jul 20 12:41:54 NOTICE[14350]: Registration from 'sip:[EMAIL PROTECTED]' failed for '69.142$ Jul 20 12:41:58 NOTICE[14350]: Peer 'dnicol' is now REACHABLE! Now, I understand the unreachable means he's gone over his qualify time, but why the registration failed *every* time after this happens? The audio always comes back after about 20 seconds, however occasionally the * server will show RTP Inactivity timeout, and drop the call. Anyone ever seen anything like this before? I understand the Qualify pings are application layer, and are not indicative of bad routing or ping times, however, with ping times of 100ms across the board and no dropped packets, I would be very surprised to see the Qualify pings go over 2000ms (the default), unless something's really got the phone cranking, no? Brian D'Arcy ___ 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
RE: [Asterisk-Users] 7960 Dynamic DNS?
I had a Netgear WGR614 802.11g Wireless Router for a short time period, it did support automatic dyndns updates, which was very handy. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 12:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? I can't think of any router that supports this You could put it in as a request to www.sveasoft.com for their firmware for the wrt54g (great box...runs linux and lots of features and functionality). P -Original Message- From: Lyle Giese [mailto:[EMAIL PROTECTED] Sent: Sunday, July 18, 2004, 9:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7960 Dynamic DNS? There are many dyn dns clients for Windoze availible and some for linux based computers. A few SOHO NAT routers support this also, but they are limited in scope and may not work for your situation. I think a workstation based solution is what you need if your router does not support it. Lyle - Original Message - From: Marty Mastera [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 8:15 PM Subject: RE: [Asterisk-Users] 7960 Dynamic DNS? snip Does anyone have any ideas on how to accomplish a dynamic dns registration without relying on a PC to do it? My router (Dell TrueMobile 2300) doesn't seem to offer this feature either. Marty ___ 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 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 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 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] IRQ Misses and Dropped Calls?
Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never had any dropped calls, or any problems. I'm receiving the following messages Jul 2 09:30:03 NOTICE[19475]: PRI got event: 4 on Primary D-channel of span 1 Jul 2 09:30:03 WARNING[19475]: No D-channels available! Using Primary on channel anyway 24! Jul 2 09:30:20 NOTICE[19475]: PRI got event: 5 on Primary D-channel of span 1 Jul 2 09:30:41 WARNING[19475]: PRI: !! Got a UA, but i'm in state 1 In between the D-Channel error notices/warings, I'll see channels 1-23 goto yellow alarm state, then I'll see them clear. It does not seem to coincide with the ~hourly reset of the b channels. I've looked everywhere for what these errors could mean, but I'm coming up empty handed. Could these errors be related to the IRQ misses I'm seeing? I'm only logging about 8 misses a day total. Zaptel and Zapata configs pasted below... [Zaptel.conf] span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us [Zapata.conf] [channels] context=inbound switchtype=dms100 overlapdial=yes signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes transfer=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group = 1 channel = 1-23 Any tips, tricks or debugging methods anyone could provide would be extremely helpful! I'm running CVS-HEAD 7/2 for libpri, zaptel and asterisk, however the problem has been occurring since we took the system live in mid-June. Thanks in advance to anyone who might can shed some light. Brian D'Arcy ___ 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] IRQ Misses and Dropped Calls?
Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never had any dropped calls, or any problems. I'm receiving the following messages Jul 2 09:30:03 NOTICE[19475]: PRI got event: 4 on Primary D-channel of span 1 Jul 2 09:30:03 WARNING[19475]: No D-channels available! Using Primary on channel anyway 24! Jul 2 09:30:20 NOTICE[19475]: PRI got event: 5 on Primary D-channel of span 1 Jul 2 09:30:41 WARNING[19475]: PRI: !! Got a UA, but i'm in state 1 In between the D-Channel error notices/warings, I'll see channels 1-23 goto yellow alarm state, then I'll see them clear. It does not seem to coincide with the hourly reset of the b channels. I've looked everywhere for what these errors could mean, but I'm coming up empty handed. Could these errors be related to the IRQ misses I'm seeing? I'm only logging about 8 misses a day total. Zaptel and Zapata configs pasted below... [Zaptel.conf] span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us [Zapata.conf] [channels] context=inbound switchtype=dms100 overlapdial=yes signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes transfer=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group = 1 channel = 1-23 Any tips, tricks or debugging methods anyone could provide would be extremely helpful! I'm running CVS-HEAD as of this morning, however the problem has been occurring over the last week, with my source being updated almost daily. Thanks in advance, Brian D'Arcy ___ 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
RE: [Asterisk-Users] IRQ Misses and Dropped Calls?
Chris, I'm afraid I can't change motherboards. It's a brand new IBM x345 2u server. I spec'd out this box specifically for asterisk, based on feedback from the community. It had all the characteristics asterisk plays well with. I'm really out of ideas here. From: [EMAIL PROTECTED] on behalf of C. Maj Sent: Fri 7/2/2004 6:31 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IRQ Misses and Dropped Calls? On Fri, 2 Jul 2004, Brian D'Arcy waxed: I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. 8's Can you try changing motherboards ? Just a guess, but it seems like you've already made it quite a few steps. Might be time to blame some hardware. --Chris -- Chris Maj, Rochester cmaj_at_freedomcorpse_dot_com Pronunciation Guide: Maj == May ___ 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 winmail.dat
RE: [Asterisk-Users] Invalid Extensions -- More like traditional PBX systems?
Steve,
What I had problems using 'i' also, what worked for me was the
following...
For example, I have a bunch of extensions I'm matching on in a
particular context. Keep in mind, the stdexten macro you see below is
not defined properly, and is for example only.
[extensions]
exten = 3201,1,Macro(stdexten)
exten = 3202,1,Macro(stdexten)
exten = 3203,1,Macro(stdexten)
exten = 3204,1,Macro(stdexten)
exten = 3205,1,Macro(stdexten)
exten = 3206,1,Macro(stdexten)
exten = _,1,Answer
exten = _,2,Wait(1)
exten =
_,3,Playback(/usr/src/test/asterisk-sounds/sounds/jedi-extension-tri
ck)
exten =
_,4,Playback(/usr/src/test/asterisk-sounds/sounds/please-try-again)
exten = _,5,Read(NUMBER,,4)
exten = _,6,Goto(${NUMBER}|1)
What happens is, after all of your possibly matchs have failed (since it
matches top to bottom in the context), it catches the _ (match all)
extension, which I consider to be invalid, since it matched none of my
defined entries above it.
From there you can do whatever you want with the call. In my case, I
decided to have some fun. =)
Brian D'Arcy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen
Rosebush
Sent: Wednesday, June 16, 2004 10:45 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Invalid Extensions -- More like
traditional PBX systems?
Yes but I try that and it doesnt even go to it, I am trying to have the
invalid handler be executed when the extension a user tries to dial from
the SIP phone is not in any of the contexts (non-existant)... I've tried
this and placed it in several contexts and it does not work.
Steve
Rob Fugina wrote:
On Wed, Jun 16, 2004 at 11:35:45AM -0300, Ray Burkholder wrote:
Set up a general pattern match with the message and congestion.
Extension pattern matching looks for the most specific match in any
one
context. So if a specific extension is not found, it will take the
general pattern.
Wouldn't it be a little better to use the special 'invalid extension'
extension -- 'i'? Something like...
exten = i,1,Playback(invalid)
exten = i,2,Busy
Rob
--
Stephen Rosebush,
[EMAIL PROTECTED]
http://www.desynched.org/
// Hardline // IP
Phone
USA:1-248-724-4452 x201 FWD:
63420 x201
Netherlands:+31-(0)20-6598858 x63420 x201 IAXTEL:
1-700-356-6191 x201
United Kingom: +44-(0)870-3403054 x201 SIP:
sip:[EMAIL PROTECTED]
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RE: [Asterisk-Users] Invalid Extensions -- More like traditional PBX systems?
Cvs checkout asterisk-sounds =) Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of W. Kevin Hunt Sent: Wednesday, June 16, 2004 12:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Invalid Extensions -- More like traditional PBX systems? exten = _,3,Playback(/usr/src/test/asterisk-sounds/sounds/jedi-extension-tri ck) exten = _,4,Playback(/usr/src/test/asterisk-sounds/sounds/please-try-again) I'd love to get a copy of that jedi-extension-trick sound W. Kevin Hunt CCIE #11841 ___ 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 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
RE: [Asterisk-Users] UIP200
Ryan, What firmware are you running on the phones? Also, are you using PoE? I have a few Uniden's I've tested with, and have experienced none of these issues. I'm running 4.54c firmware, and not using PoE. I currently have one of these phones in production at a remote location. It gets heavy use and displays none of the reboot/disconnect issues you've described. You should be able to fix the DTMF/IVR problem by using dtmfmode=inband for each peer in your sip.conf. If you'd like the 4.54c firmware (I got it from uniden directly, as it fixes a DHCP issue they have), e-mail me directly and I'll be happy to provide it. Maybe it'll fix some issues you're having. Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: [EMAIL PROTECTED] Web: http://www.akiva.com Phone: 760-710-3209 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage Sent: Wednesday, June 16, 2004 2:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] UIP200 Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at the remote end). Interactions local to * (ie: vmail) do work (using rfc2833). I've been able to duplicate these problems both on a customer site, and in my own test environment. Has anyone else using the UIP200 experience any of these issues? Still waiting for Uniden Support to return my call. Thanks .. Ryan Courtnage Coalescent Systems 403.244.8089 www.voxbox.ca ___ 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 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
RE: [Asterisk-Users] Cisco 7970 w/ 7.1 phones rebooting with asterisk
Sorry, also running 7.1 and Head as of 6/8/04. No reboot problems here at all with voicemail. If you're using some sort of custom onscreen voicemail application, there are known issues with improperly formatted XML causing the phones to either lock up, or reboot. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, June 10, 2004 2:48 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7970 w/ 7.1 phones rebooting with asterisk I am currently testing an asterisk server with some cisco 7960 phones. I have been having problems with phones rebootin using 6.3 firmware in the asterisk voicemail menus. The phones reboot after a dozen or so random button presses while in the voicemail menus. To try and fix this, I upgraded to sip 7.1 only to find that now the phones reboot even if i'm trying to press a button to dial out. I am running CVS head as of 2 days ago. If anyone has ever had any problems like this please let me know Nathan ___ 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 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] No ringing on outbound PRI calls
Ive got a strange issue where I get no ringing on outbound PRI calls using the TE410P span 1. The call actually goes out and works, you just hear no ringing. The quality and features on the call are pristine, no cracks pops or any weirdness like that. If I specify the r option on the Dial() string, I get a half of a ring, sometimes a full ring as soon as the dial completes. Ill paste my configs below incase someone can see somethings that set wrong. Any help anyone can provide would be greatly appreciated! Zaptel.conf and Zapata.conf are configd correctly (identically do the PBX which * replaced just today, no problems with ringing on the old pbx, same PRI). Im running head for zaptel, libpri and asterisk as of 06/08/04. [zaptel.conf] span=1,1,0,esf,b8zs bchan=1-23 dchan=24 defaultzone=us loadzone=us [zapata.conf] [channels] context=inbound switchtype=national overlapdial=yes signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default group = 1 channel = 1-23 Thanks! Brian D'Arcy
[Asterisk-Users] Cisco 7960 XML/Configs
I ordered 10 7960s with SIP today (YAY!), I should have them on Monday! So, to be better prepared come Monday morning, I was wondering if anyone knew of any * compatible screen configs for things such as browsing VM, etc, yadda, yadda. I checked out the wiki about ADSI but from what I see, thats not really applicable in a SIP setup? Im guessing its going to be a more XML and static HTML based type of setup. If anyone can point me to some resources or has some scraps of examples laying around, Id love to take a look, as Im really stoked about doing some cool stuff with these phones and *. =) Have a nice weekend everone! Brian D'Arcy
[Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Hello all,
I'm going to tackle learning C this week, and start writing my first *
add-on/contribution; assuming it's actually worthy of contributing once
it's done.. I think I've chosen a hefty project for my first go round
here...
I'd like to get some feedback from everyone on a FindMe/FollowMe spec
I've put together. Before you read on, let me say, I don't want this to
turn into a it would be cool if it did this.., or that etc... I'm
writing this to serve a very simple and basic function, and I want it to
do exceedingly well at just that for starters.
Please check out specs below as to how I envision it working within a
dialplan environment, and also, please keep in mind this is being
written to be used in a corporate environment. There are a lot of
others out there with far more * experience than myself, so any
constructive criticism would be most welcome as to the layout and
configuration of the soon to be app_findme.
Thanks!
Spec for app_findme
Have a .conf file (findme.conf?) which contains multiple contexts, each
context's name should match the naming convention used with sip, or
iax.conf. For example, if I have [bdarcy] as one of my sip peer
entries, in findme.conf I would have, [bdarcy] also listed as an entry.
Values within each entry would be labeled something like,
[bdarcy]
ExternalNum1: 91235551212
ExternalNum2: 91235551213
etc...
app_findme would be used as the unavailable behaviour within the
dialplan (or could be used in both unavailable and busy), for example
[macro-stdexten]
exten = s,1,Wait(1)
exten = s,2,Dial(${ARG2},20,tTr)
exten = s,3,FindMe(${ARG2})
exten = s,4,Voicemail(u${ARG1})
exten = s,5,Wait(4)
exten = s,6,Hangup
exten = s,104,Voicemail(b${ARG1})
exten = s,105,Wait(2)
exten = s,106,Hangup
As the default unavailable behaviour, it always tries the findme
application, if no entries for this person exist in findme.conf, it
continues on in the dialplan, and hits the unavailable voicemail. If
entries are found:
Call gets answered, caller hears:
Hello, please wait while I try and find the person you are calling.
(MOH)
Every 10 seconds play to the caller: Still trying to find this person,
please wait..
Callee answers, app_findme says: There is a call for you from (CIDNum),
to accept this call, press *, otherwise press #, or hangup.
If I press *, the caller hears, I have found this person, connecting you
now..
Caller hears: I have found this person for you, connecting you now..
If # is pressed, the callee hangs up, or it never receives the *
confirmation tone, the caller hears: Sorry, I was unable to find this
person for you. and +101's the priority sending them into the busy
voicemail.
I look forward to hearing back from everyone on this. I'm really
excited to start learning, and feedback from the community will help
motivate me, while also ensuring I don't shelve this project just to
play some XBOX and drink some beer during my free time!
Brian D'Arcy
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RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Thanks Andrew, good feedback. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Tuesday, June 01, 2004 2:22 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec. Brian D'Arcy wrote: Spec for app_findme Have a .conf file (findme.conf?) which contains multiple contexts, each context's name should match the naming convention used with sip, or iax.conf. For example, if I have [bdarcy] as one of my sip peer entries, in findme.conf I would have, [bdarcy] also listed as an entry. Values within each entry would be labeled something like, [bdarcy] ExternalNum1: 91235551212 ExternalNum2: 91235551213 etc... snip Call gets answered, caller hears: Hello, please wait while I try and find the person you are calling. (MOH) Every 10 seconds play to the caller: Still trying to find this person, please wait.. Callee answers, app_findme says: There is a call for you from (CIDNum), to accept this call, press *, otherwise press #, or hangup. If I press *, the caller hears, I have found this person, connecting you now.. Caller hears: I have found this person for you, connecting you now.. If # is pressed, the callee hangs up, or it never receives the * confirmation tone, the caller hears: Sorry, I was unable to find this person for you. and +101's the priority sending them into the busy voicemail. All of the voice streams outputted should be defined in the [general] section of the .conf file. They should also be user-overridable on a per peer(?) or per context basis. - Andrew Thompson http://aktzero.com/ ___ 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 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
RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Hi Tony, I just reviewed the privacy feature you linked me to. While this is similar, it's really a totally separate beast for a few reasons. The goal of my app is to provide a very simplistic way to have the server track you down at multiple phone numbers. When using asterisk as a home system, for one or two extensions, yeah this would be cake in the dialplan. However, I'm dealing with replacing a large commercial pbx, where we have 40 employees using this FindMe Feature. The idea of having an external conf file used for defining each employees findme numbers will help keep my already 1000+ line extensions.conf a little more sane. Especially since *all* extensions go through the standard dial macro. Without using the separate config file, I would have to have *multiple* dial statements, each reflecting each employees find me phone number(s). Yuck. The way I see it, you've got sip.conf, iax.conf, and voicemail.conf. everything is already separated, so there's no real reason to junk up the dialplan any more than it is (100 did's + 40+ extensions + IVR's + international calling, local calling etc..), so why not make a matching configuration file for external phone numbers or devices. I will however play with this new privacy feature, as it will undoubtedly help me in my coding efforts, being a newb and all. =) Thanks for your feedback, Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Tuesday, June 01, 2004 2:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec. Brian D'Arcy wrote: I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want this to turn into a it would be cool if it did this.., or that etc... I'm writing this to serve a very simple and basic function, and I want it to do exceedingly well at just that for starters. Hi Brian, I've been looking for something like this and I think your efforts might overlap those of the work-in-progress Privacy option for app_dial. See http://bugs.digium.com/bug_view_page.php?bug_id=752 It would seem that if you were to instead add one more flag to the Dial app, that places calls in sequence rather than all at once, you would have everything you need. Perhaps it might even be better to work with Steve Murphy and turn his app_dial patch into a stand-alone app_followme? I just see an overlap here and you probably don't want to reinvent the wheel :) Regards, Trevor Peirce ___ 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 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
RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec.
Oops.. that'd be Trevor, not Tony. My fault =) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor Peirce Sent: Tuesday, June 01, 2004 2:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Feedback needed! FindMe/FollowMe Feature Spec. Brian D'Arcy wrote: I'd like to get some feedback from everyone on a FindMe/FollowMe spec I've put together. Before you read on, let me say, I don't want this to turn into a it would be cool if it did this.., or that etc... I'm writing this to serve a very simple and basic function, and I want it to do exceedingly well at just that for starters. Hi Brian, I've been looking for something like this and I think your efforts might overlap those of the work-in-progress Privacy option for app_dial. See http://bugs.digium.com/bug_view_page.php?bug_id=752 It would seem that if you were to instead add one more flag to the Dial app, that places calls in sequence rather than all at once, you would have everything you need. Perhaps it might even be better to work with Steve Murphy and turn his app_dial patch into a stand-alone app_followme? I just see an overlap here and you probably don't want to reinvent the wheel :) Regards, Trevor Peirce ___ 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 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
RE: [Asterisk-Users] Feedback needed! FindMe/FollowMe FeatureSpec.
Hi Adam, I appreciate your feedback, and understand totally where you're coming from as far as the database perspective goes. For the first draft of the app, I think I'm going to let it default to using a conf file for two reasons. First, my setup currently does not utilize a database. I would like to move to that type of a setup in the future however. Secondly, seeing as how I'm sitting down to learn C this week, I think that is definitely biting off more than I can chew for a first attempt! Should my endeavors pan out, I would be more than happy to implement odbc connectivity. I need to become more familiar with post and mysql first however. Up to this point, I've been strictly a MSSQL DBA due to my job functions. Thanks again for your feedback. Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: [EMAIL PROTECTED] Web: http://www.akiva.com Phone: 760-710-3209 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: Tuesday, June 01, 2004 5:50 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Feedback needed! FindMe/FollowMe FeatureSpec. On Wed, 2004-06-02 at 06:14, Brian D'Arcy wrote: Hello all, Have a .conf file (findme.conf?) which contains multiple contexts, each context's name should match the naming convention used with sip, or iax.conf. For example, if I have [bdarcy] as one of my sip peer entries, in findme.conf I would have, [bdarcy] also listed as an entry. Values within each entry would be labeled something like, [bdarcy] ExternalNum1: 91235551212 ExternalNum2: 91235551213 etc... Good idea, I encourage you to go ahead and do this! However, please don't use a seperate .conf file. Either use the dbodbc interface (if there is one) or at least use the astdb interface. If someone 'forgot' their mobile, but is going to be sitting at their friends house for the day, they 'should' be able to call into the asterisk pbx, enter some dialplan ext/password stuff, and remove/turn off the mobile followme step, and add in the local phone. Should also be able to set a priority for each followme number, so they are tried in order of your preference. To store in the db something like: /followme/6600_1: 90402xx /followme/6600_2: 6654 /followme/6600_3: 98424 This would represent the following attempts to connect the call: First, try me on my mobile, based on the phone number, (starts with a 9) it should grab a line in zap/g2 and call 0402xx If that fails, then try me on a local extension (starts with 6) 6654 If that also fails, then try me on a landline (starts with 9) so grab a line from zap/g2 and call 8424 I don't know if it is possible to 'use/abuse' the dialplan in this way, where you can sort of follow the dialplan for a 'while' and then after the dial command fall-back to the app and allow it to continue. The other option if that isn't possible is to specify the channel details in the astdb like this: /followme/6600_1: Zap/g2/0402xx /followme/6600_2: Zap/124 /followme/6600_3: Zap/g2/8424 Actually, on second thoughts, couldn't we just use the Local channel driver like this: Using the first db above, we need to call 90402xx so we call: Dial(Local/90402xx) which should follow the dialplan to make the call, and if it fails come back to us Perhaps comments on the above from other people would help. I think trying to manage a conf file for a few hundred people would be almost as bad as trying to put this stuff into the extensions.conf Although, extensions.conf sounds like it will be accessible from a DB shortly, so this might in fact be your better option anyway Just my 0.01c worth... Regards, Adam ___ 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 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
RE: [Asterisk-Users] Asterisk Receptionist manager program.
Kyle, I also would be very interested. It may negate the purchase of a much more expensive phone in the future. =) Thanks, Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Friday, May 28, 2004 9:33 AM To: Asterisk Subject: [Asterisk-Users] Asterisk Receptionist manager program. We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. We are considering writing a SIP client build into the program at a later time. Kyle ___ 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 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
RE: [Asterisk-Users] spandsp hylafax asterisk and confusion
I actually got mine compiled this morning.. (worked most of the day on it yesterday). I made the makefile changes to app_rxfax, app_txfax, and app_dtmftotext someone in the list recommended, as well as the modification to the lock.h in the includes directory. Everything compiles peachy then.. However when I start asterisk... [app_rxfax.so]May 25 08:43:21 WARNING[1024]: loader.c:240 ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory May 25 08:43:21 WARNING[1024]: loader.c:421 load_modules: Loading module app_rxfax.so failed! asterisk:/usr/src/asterisk# Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Ouch ... error while writing audio data: : Broken pipe Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Junk at the beginning 49443303 Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe I can handle the trial and error as far as getting something setup and compiled, but I think now that I've made it this far it's out of my hands.. I haven't the slightest on where to start now.. Yes, I'm running CVS-HEAD-05/24/04-16:20:41 Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: [EMAIL PROTECTED] Web: http://www.akiva.com Phone: 760-710-3209 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Tuesday, May 25, 2004 7:38 AM To: Terry Goodwin Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp hylafax asterisk and confusion On Tue, 25 May 2004, Terry Goodwin wrote: Thanks for offering to help with this. I checked out the procedures and attempted this again without success. Here is the end of the screen output when the compile fails. gcc -02 -g -Include -I ../include -c -o app_rxfax.o app_rxfax.c app_rxfax.c:45: error: 'PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP' undeclared here (not in a function) make[1]: *** [app_rxfax.o] error 1 make[1]: leaving directory '/usr/src/asterisk/apps' make: *** [subdirs] Error 1 Ah - I remember this. There may be other fixes, but I resolved this by adding: #ifndef _GNU_SOURCE #define _GNU_SOURCE #endif Just before the #include pthread.h in asterisk/include/asterisk.lock.h Regards, Steve Davies ___ 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 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
RE: [Asterisk-Users] spandsp hylafax asterisk and confusion
Thanks everyone for your responses. While these tips and tricks did infact help get asterisk compiled with the fax modules, it seems that * still craps out on the app_dtmftotext.c when you first start it. I can't seem to find a way to get rid of it. I'm not even totally sure it's required to send or receive faxes. If anyone has a step by step (more like, location by location) as a work around for that, I'd be all ears. I thought removing the lines in the Makefile for app_dtmftotext.c would be enough for it to be excluded, but apparently it's not. If it's this much of a pain to get the fax modules installed everytime I update from CVS, it makes me wonder if the $8/mo I pay to JFAX isn't worth it! =) Cheers, Brian D'Arcy Operations Engineer Akiva Corporation E-Mail: [EMAIL PROTECTED] Web: http://www.akiva.com Phone: 760-710-3209 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wade J. Weppler Sent: Tuesday, May 25, 2004 2:44 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] spandsp hylafax asterisk and confusion Or just add /usr/local/lib to your /etc/ld.so.conf file. -wade -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaus Darilion Sent: Tuesday, May 25, 2004 1:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] spandsp hylafax asterisk and confusion Brian D'Arcy wrote: ast_load_resource: libspandsp.so.0: cannot open shared object file: No such file or directory I copied the libspan* files from /usr/local/lib to /usr/lib and then asterisk started! klaus ___ 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 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 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
RE: [Asterisk-Users] IAXy
Title: Message If you just want a test unit, goto www.netxusa.com. Thats where digium sent me if I wasnt ordering in bulk. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, May 13, 2004 2:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAXy Not sure if this is the best place but does any one have any used IAXy's they are interested in selling? I am looking topick one up cheap for a proof of concept before going all out on them. Also does any one have any real life practical experience with how well (or not so well) that these devices have worked for them? you can reply to me off list at [EMAIL PROTECTED] Thanks Michael Blood
[Asterisk-Users] Virbiage FT201 IAX Hard Phone
Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... Thanks, Brian D'Arcy ___ 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
RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone
Uniden does have a SIP phone. I posted a review of it on Friday of last week. There were some problems with the list last week, so look for a re-post of the review, as well as contact information for a distributor in a few moments. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of brian Sent: Monday, May 10, 2004 12:18 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone The response from them less than 10 days ago was 'about 2 months'. Which is what they have been saying for the last *3* months. I'm giving up waiting. They seem to be vaporware. :( Does anyone have a supplier for the Uniden UIP200 phones? uniden's are all h323 last I checked. ( h.323 aka HELL 323 IMHO ) :P bkw ___ 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 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] Uniden UIP200 Review (Repost)
Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some feedback from an office perspective on the new unit. It is Uniden's first offering into the VoIP market. Main Features which were important to me: Built in 10/100 Switch Speakerphone w/headset port IEEE 802.3af Standard Inline Power (PoE) 2 line 16/char LCD Display 8 Programmable (not soft) Keys QoS [IEEE 802.1 p/q Based and DiffServ G711a/u G729A Codec Support TFTP Auto Configuration Firmware Upgrades (based on mac addressed filenames) The phone also has all the hard buttons you'd expect it to have. Hold, speaker/headset, Volume up and down, Menu, Transfer, Cancel, and Dial (used in lieu of pressing the # key to cut down digit timeouts when on-hook dialing). First, this phone, is relatively inexpensive. I was able to pick one up for $129. Setup and configuration was trying, as the phone ships with absolutely NOTHING in terms of an admin guide. The support areas on the Uniden site were password protected and even the support staff was unaware of all the proper logins and passwords (gotta love supporting new products). Once I gained access to the appropriate admin guide, I whipped up a few of the configuration files on my TFTP server, plugged in the phone and was off and rolling. Or so I thought. There seems to be some minor DHCP issues with the phone currently. It was ignoring my DHCP server's DHCP Offer's and constantly reported DHCP Failed on the LCD. After speaking with a Uniden Developer and sending him an ethereal trace, I hard-coded the IP address to continue my testing. The phone fired up, auto-configured itself via TFTP, and was logged into * in a matter of seconds. Needless to say, at this point, I was extremely pleased to see it actually WORKED. Weak Points: Wimpy Speakerphone: It's extremely easy for the speakerphone itself to over modulate. The microphone however does seem to perform well, even if it is a *little* tin-can'ish. Hold Button: Works as expected, * puts the caller on hold, and they hear MOH. YOU on the other hand hear this really cheesy Nintendo style genre of music locally, produced by the phone. When using speakerphone and placing someone on hold, this is extremely irritating. DTMF: When you have a session, or call active, there is no local DTMF feedback over the handset or speakerphone. While I'm ok with this, I can just picture my entire office on the first day, wondering if they actually pushed the buttons hard enough. So navigating through auto attendant menus can be a little tricky since you're not sure if you actually missed the button, or made solid contact. You can however check the LCD to see if the number you pressed went through. Conclusion: In testing, the phone is an all around solid performer. If they resolve my DHCP issue, I think we probably will go ahead and purchase 20-30 phones to start so that we can get * deployed in the near future. For $130, I don't think I can really complain about the weak points, however I have voiced my opinion on the DTMF and HOLD music to Uniden, so maybe in the near future we'll have some toggles in the TFTP config files make life a little less stressful. Uniden currently has a distributor/wholesaler who will sell to the public. If you're interested in picking up any of these phones to test yourself, the contact information is below. Note: Please keep in mind, Uniden also makes the UIP300 and UIP312. These phones *only* support H323. The UIP400 is the equivalent model of the 300, but will support SIP and is currently in development. Contact: Aimee @ Teledynamics (800) 847-5629 ext.110 or, [EMAIL PROTECTED] Brian D'Arcy ___ 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
RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone
Thanks for the reply Tim. That's a real shame. The featureset of the soon-to-maybe-be Virbiage IAX hardphones seemed too good to be true. I guess I'll go ahead with a full SIP deployment, and hope IAX hardphones become a reality in the future. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Monday, May 10, 2004 11:35 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone On Mon, May 10, 2004 at 10:53:14AM -0700, Brian D'Arcy wrote: Does anyone have any recent news on the Virbiage FT201 IAX Hardphone? I'd *really really* like to deploy these phones instead of SIP hardphones, and I can't help but wonder if I'm going to shoot myself in the foot (or another sensitive area) by deploying a ton of SIP phones just to find the IAX Hardphones were released a week later... The response from them less than 10 days ago was 'about 2 months'. Which is what they have been saying for the last *3* months. I'm giving up waiting. They seem to be vaporware. :( Does anyone have a supplier for the Uniden UIP200 phones? Tim -- Tim Sailer Coastal Internet, Inc. Network and Systems Operations PO Box 726 http://www.buoy.comMoriches, NY 11955 [EMAIL PROTECTED] (631) 399-2910 IAX 17003992910 ___ 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 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
RE: [Asterisk-Users] Virbiage FT201 IAX Hard Phone
Michael, Ive heard Todds name also, however a few people from the * IRC room gave him a call and were told that they would not be sold to. Ive had the same perception and experience with their staff. Extremely professional and go out of their way to make you happy. A great company to do business with (Teledynamics). An additional contact can be found in the review of the UIP200 post which should have just gone up a few minutes ago. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Swan Sent: Monday, May 10, 2004 12:53 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Virbiage FT201 IAX Hard Phone At 02:34 PM 5/10/2004 -0400, you wrote: The response from them less than 10 days ago was 'about 2 months'. Which is what they have been saying for the last *3* months. I'm giving up waiting. They seem to be vaporware. :( Does anyone have a supplier for the Uniden UIP200 phones? Tim -- This is the contact that Uniden gave me. The guy is super responsive and proactively sent me email when the phone was shipping. Don't know if the UIP200 is any good as I've not yet received it, but Todd is great! Todd Baca Key Accounts Manager TeleDynamics Tel: 1-800-847-5629 x 119 [EMAIL PROTECTED]
[Asterisk-Users] Uniden UIP200 Review
Hello Everyone, My company is about to deploy * as replacement for our existing commercial Altigen PBX. Meanwhile, I've been trying to find the best cost effective SIP VoIP phone which we can use in office for 20-30 employees, as well as a few remote staff. Normally I wouldn't post about a VoIP phone, however, this phone was released less than a week so I thought I'd give some feedback from an office perspective on the new unit. It is Uniden's first offering into the VoIP market. Main Features which were important to me: Built in 10/100 Switch Speakerphone w/headset port IEEE 802.3af Standard Inline Power (PoE) 2 line 16/char LCD Display 8 Programmable (not soft) Keys QoS [IEEE 802.1 p/q Based and DiffServ G711a/u G729A Codec Support TFTP Auto Configuration Firmware Upgrades (based on mac addressed filenames) The phone also has all the hard buttons you'd expect it to have. Hold, speaker/headset, Volume up and down, Menu, Transfer, Cancel, and Dial (used in lieu of pressing the # key to cut down digit timeouts when on-hook dialing). First, this phone, is relatively inexpensive. I was able to pick one up for $129. Setup and configuration was trying, as the phone ships with absolutely NOTHING in terms of an admin guide. The support areas on the Uniden site were password protected and even the support staff was unaware of all the proper logins and passwords (gotta love supporting new products). Once I gained access to the appropriate admin guide, I whipped up a few of the configuration files on my TFTP server, plugged in the phone and was off and rolling. Or so I thought. There seems to be some minor DHCP issues with the phone currently. It was ignoring my DHCP server's DHCP Offer's and constantly reported DHCP Failed on the LCD. After speaking with a Uniden Developer and sending him an ethereal trace, I hard-coded the IP address to continue my testing. The phone fired up, auto-configured itself via TFTP, and was logged into * in a matter of seconds. Needless to say, at this point, I was extremely pleased to see it actually WORKED. Weak Points: Wimpy Speakerphone: It's extremely easy for the speakerphone itself to over modulate. The microphone however does seem to perform well, even if it is a *little* tin-can'ish. Hold Button: Works as expected, * puts the caller on hold, and they hear MOH. YOU on the other hand hear this really cheesy Nintendo style genre of music locally, produced by the phone. When using speakerphone and placing someone on hold, this is extremely irritating. DTMF: When you have a session, or call active, there is no local DTMF feedback over the handset or speakerphone. While I'm ok with this, I can just picture my entire office on the first day, wondering if they actually pushed the buttons hard enough. So navigating through auto attendant menus can be a little tricky since you're not sure if you actually missed the button, or made solid contact. You can however check the LCD to see if the number you pressed went through. Conclusion: In testing, the phone is an all around solid performer. If they resolve my DHCP issue, I think we probably will go ahead and purchase 20-30 phones to start so that we can get * deployed in the near future. For $130, I don't think I can really complain about the weak points, however I have voiced my opinion on the DTMF and HOLD music to Uniden, so maybe in the near future we'll have some toggles in the TFTP config files make life a little less stressful. Uniden currently has a distributor/wholesaler who will sell to the public. If you're interested in picking up any of these phones to test yourself, the contact information is below. Note: Please keep in mind, Uniden also makes the UIP300 and UIP312. These phones *only* support H323. The UIP400 is the equivalent model of the 300, but will support SIP and is currently in development. Contact: Aimee @ Teledynamics (800) 847-5629 ext.110 or, [EMAIL PROTECTED] Brian D'Arcy ___ 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
RE: [Asterisk-Users] List of online sip users
Holger, From the Asterisk CLI, type: sip show peers This will show you all users currently registered with Asterisk. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Holger Zimmermann Sent: Saturday, May 08, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] List of online sip users Hello list, is it possible to see all online users? I have configure a isdn2sip gateway in the company I work. Now, the question: Is it possible to show all colleague which people where reachable with this gateway? greetings and thanks, Holger _ Der WEB.DE Virenschutz schuetzt Ihr Postfach vor dem Wurm Netsky.A-P! Kostenfrei fuer alle FreeMail Nutzer. http://f.web.de/?mc=021157 ___ 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 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
RE: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP
Hi Russ, Thanks for your feedback! I hadn't received any other responses from anyone, so I was starting to worry that I was one of the few having these erratic issues. I might ping Sonicwall, being a good customer and all, maybe I can get some information out of them. I've always liked using the sonicwall for ease of use and administration (and reliability), since I'm overworked as it is, but if I have to get rid of it to make this work, I'm not against it. On a side note, I tried IAX2 last night for the first time using IAXPHONE. HOLY CRAP I'M IMPRESSED!!! Everything just *works*, period. I might just use softphones until IAX hardphones are released and say screw SIP. If anyone else is having SIP nightmares and you have a flexible deployment schedule, I highly recommend giving IAX a shot!! Thanks again for the comments, Russ. Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Russ Beaupre, P.E. Sent: Saturday, April 24, 2004 5:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk configuration inside a DMZ w/SIP Brian D'Arcy wrote: Hello all, I'm having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I can't seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. We've discovered that certain versions of the sonic wall products do strange things with SIP. For example the TC170 with standard firmware works fine (Public Asterisk, Polycom IP600 behind the Sonic wall). Upgrade that box to the enhanced version and suddenly transfer and hold stop working. It's not just SIP, either. SNTP on the IP600 through the Sonic Wall gear changes the time by 10 hours. These things have been reported to Sonic Wall, but no word on a patch. -rb ___ 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 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] Asterisk configuration inside a DMZ w/SIP
Hello all, Im having a nightmare of a time trying to get stable results with SIP clients on Asterisk. I cant seem to find a configuration that works! In our office, we run a Sonicwall Pro 200, which is a sip aware, stateful firewall. Originally, I had configured Asterisk to run on the NAT side so that those within the office could connect easily, and those outside the office could connect via VPN. However the VPN route is proving to be a little too latent for quality calls. Even still, some people were able to receive audio, and others not. After much reading about Asterisk and the problems inherent to NAT, I decided OK, Ill just toss it on the DMZ with a public address, and let the clients themselves worry about addressing their NAT issues @ home, or wherever they might be. So here I am, with Asterisk running on the DMZ with a public IP address, totally unfirewalled to the outside world and now I find that not only can I not connect (from the nat side of the same SIP aware firewall hosting the asterisk server), but clients on public IPs, using no NAT at all, are either unable to connect, or are able to log in, but calls to any extension (whether they be sip extensions, voicemail, conference etc..) come up 408 timed out. In every case, the message in the * CLI is reported as: chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 30841 (Response) This to me would imply that for whatever reason, the packets from the Asterisk server are being blocked by the local firewall when it attempts to send them back to me. This I can understand, because maybe Im having NAT issues myself, however I get the *same* messages broadcast into the CLI when users on the public IP addresses attempt to connect in (unfirewalled). Ive checked and triple checked to make sure that the DMZ port is not firewalled in any way, so Im a bit stumped. After this rambling, I suppose the real question Im asking here is, what is the most stable, preferred networking setup people tend to use when they are expecting to have SIP clients connecting both internally, and externally? Incase everyone wants to know about my SIP configurations, Im using disallow=all, and allow=ulaw ONLY. Ive toyed with the nat=1/nat=yes settings, however they seem to have no real effect on the behavior of the clients. Ive been testing strictly with X-Lite, as it came recommended by a few folks in #Asterisk on irc.freenode.net. [General] section from SIP.conf and an example SIP client entry: [general] port=5060 ; Port to bind to bindaddr=0.0.0.0 ; Address to bind SIP channel to ;externip = 216.9.32.42 ;localmask=255.255.254.0 ;localnet=192.168.0.0 context = default ; Default context for incoming calls ;srvlookup = yes [bdarcy] type=friend username=bdarcy secret=blah host=dynamic qualify=400 mailbox=3209 callerid=Brian D'Arcy 3209 nat=1 disallow=all allow=ulaw If anyone can provide any feedback on what works for you, or whats recommended, it would be highly appreciated. Thanks in advance. Brian D'Arcy
