[asterisk-users] Forward incoming call to recipients.
Hey, all. I'd like to forward an incoming call (e.g., to an on-call rotation number), out to multiple recipients, BUT only hand the call over to whoever answers _and acknowledges_ (e.g., "Press any key..."), 'cause I don't want it just going to their mailbox. I've thought of a number of ways to try to make this happen, but surely, something like this must be fairly common, and I'm guessing someone's already got The Right Way(tm) to make it happen. Any suggestions? Thanks! -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM cards
On 2020-02-16 16:48, Dovid Bender wrote: Hi, It's been forever since I dealt with POTS lines. We have a client that needs FXS and FXO support. If memory serves correct we used the TDM400P with fxs_gs/fxo_gs. What's the equivalent of that card today? Wow. Lotta replies. I haven't done POTS in forever, either, but when I last did, it was with the Sangoma cards, like so: https://www.sangoma.com/telephony-cards/analog/ I normally wouldn't post Sangoma to a Digium list... but since Sangoma went and acquired Digium, I guess it kinda makes sense. -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] headers in master.csv
On 2018-04-26 09:49, John Tuxies wrote: Hi. i am looking for a way to have headers for each section of the Master.csv eg call duration, hangup cause, destination,... is there a way to add it and be there permanently, even after log roratation due to size or date, please? That's easy: no. ;-) But, less-snarkily, that's the kind of thing that scripting is made for. Clearly, you're looking to import it into spreadsheet. In your shoes, I'd set up a file with *just* the headers, and then when you wanted a CSV for import, do something like this: cat heaerfile.txt master.csv > file_to_import.csv And lo! Done. -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom and forwarding.
Hey, all. I've got an office set up with Asterisk, and forwarding's got a bit of a glitch: When they forward, they listen for the remote phone to ring, then hang up. If the remote phone doesn't connect, it goes to the original phone's VM. Is this Polycom's fault, or Asterisk's? I've been reading up on blind/supervised forwards, and, honestly, have myself more confused than when I started. If someone could give me a solid idea of how forwarding works, and a sample of how to send it to a remote extension, and have it *not* come back to the original extension, that'd be great. Thanks, -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dahdi-channels.conf vs. chan_dahdi.conf
Hey, all. Just added an analog card to our dual-T1 system... and clearly I'm doing something wrong. Less interested in having the specifics pointed out than in finding out how/why certain things work. So, really, three things: * What the bloody Hell is the difference between dahdi-channels.conf and chan_dahdi.conf? (And who thought it was a good idea to have two files with, apparently, different functionality, but very similar names?) * If I'm getting power to my analog phones, but no dial tone, which file should I be editing? * Likewise (and almost certainly related) if dahdi_cfg shows the channels, but dahdi show channels only shows my T1 spans, which file should I be editing? Could someone point me to some sample analog configs? Most of my searches have wound me up with GUI folks, and I'm just doing good ol-fashioned hand editing on an Ubuntu system. Thanks! -Ken -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Desktop SIP phone with OpenVPN client
Not sure why you'd say it's OT -- seems perfectly topical to me. Anyway, I have used the SNOM OpenVPN feature for remote clients. I'll be honest: it's a bit of a pain to set up and get working. This is triply true if the remote phone is moving around -- to the point that I'd strongly advise against it. But if it's staying in one place (e.g., a telecommuter's desk) once it's at the remote site, it should do the job, and reasonably well. It certainly can use DHCP, though I rather doubt it needs it -- not quite sure what you're angling for by way of your question. WiFi was a PITA; I'd advise against. (Esp. since -- at least, at the time -- you needed to download beta firmware to get both WiFi and OpenVPN working at the same time.) Note that I haven't touched this in about two years' time, so things may have changed a bit in the interim. -Ken On 2013-01-21 08:21, Olivier wrote: Hello, I've seen some desktop SIP phones (Snom, Yealink) intregrate a VPN (OpenVPN ?) client. Has someone experience to share about that particular feature ? Is this experience rather successful ? My underlying question is can one supervise and configure these desktop phones, in teleworking environment ? Is DHCP required ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Minimal pass-through T1 configuration?
Hi, all. I'm upgrading my company's old 1.2 box with a new-and-improved one. But a fair bit's changed in the interim. To start, at least, I just want the new box to act as a pass-through for all calls -- PSTN calls go, unmodified, to the internal T1, and vice-versa. (That way, I can begin to build a skeleton dialplan and work my way forward from there.) But I'm bumping into problems: -- Executing [6000@pstn:1] Dial(DAHDI/i1/6034941234-1, DAHDI/G2/6000) in new stack [Jan 21 12:19:35] WARNING[15163][C-]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) So, a few questions: 1) What's the difference between chan_dahdi.conf and dahdi-channels.conf? 2) Could someone show me a SuperDumb(tm) minimal config for the necessary /etc/*dahdi*.conf and /etc/extensions.conf files? Thanks! -Ken -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf
type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf
#include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf
type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN})
exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that point
from SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=172.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 ---
ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm=asterisk,
nonce=16883b72, uri=sip:7444@172.17.0.17,
response=b75389c5938b4f185b3d31bd4463abf3, algorithm=MD5
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken
D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between
two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include = aggregate
include = passthrough
exten = _7XXX,1,Dial(SIP/box2/${EXTEN}) exten = _7XXX,2,Hangup()
---
When I dial, all I get is (I'll attach the full dialog up to that
point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial(SIP/6110-08291cb0,
SIP/box2/7444) in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
---
Where am I goofing up? Any pointers?
Thanks!
-Ken
---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4
172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
-
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk,
nonce=16883b72
Content-Length: 0
Scheduling destruction of SIP dialog
'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
--- SIP read from 172.17.9.1:55388 --- ACK sip:7444@172.17.0.17
SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: sip:172.17.0.17;transport=udp;lr
Content-Length: 0
-
--- (9 headers 0 lines) ---
--- SIP read from 172.17.9.1:55388 ---
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: sip:6110@172.17.0.17;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: sip:7444@172.17.0.17
Contact: sip:6110@172.17.9.1:55388;ob
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: sip:172.17.0.17;transport=udp;lr
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username=6110, realm
[asterisk-users] Trunking through an old Asterisk box.
Hi! I'm helping set up a new Asterisk box. However, since I can't just take the T1 to play with (and I *will* be making many changes, e.g., going to Adhearsion), in order to test my dialplan, I'll need to route calls through the old, Asterisk 1.4 box. I've never really done this. What's the right way to go about it? Thanks, -Ken -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI and AMI stuff.
Hey, all. I'm interested in doing some simple, very specific web pages for some of my users -- things like call groups, setting forwarding, and for the receptionist to transfer calls and see calls. Probably do this in Ruby or PHP, though I'm open-minded. Anyway, if someone could point me to some documentation -- dead tree, electronic, whatever -- that gives some fairly in-depth detail on this, I'd be most appreciative. Thanks! -Ken -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI and AMI stuff.
Heh. Shortly after I sent my e-mail, I bumped into the Adhearsion you mentioned, below. Boy, but that looks exactly like what I'm thinking of! Thanks much... -Ken On 2012-11-15 13:08, David M. Lee wrote: On Nov 15, 2012, at 10:54 AM, Ken D'Ambrosio wrote: Hey, all. I'm interested in doing some simple, very specific web pages for some of my users -- things like call groups, setting forwarding, and for the receptionist to transfer calls and see calls. Probably do this in Ruby or PHP, though I'm open-minded. Anyway, if someone could point me to some documentation -- dead tree, electronic, whatever -- that gives some fairly in-depth detail on this, I'd be most appreciative. AGI commands and AMI actions and events are documented on the wiki - https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Command+Reference [1] The wiki also has command references for 1.8 and 10. Asterisk: The Definitive Guide has good chapters on the protocols: * Book - http://shop.oreilly.com/product/9780596517342.do [2] * AMI - http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html [3] * AGI - http://ofps.oreilly.com/titles/9780596517342/AGI.html [4] There are client libraries that handle the protocol details for you. StarPy for Python is a fairly low-level wrapper around AMI/AGI. Adhearsion for Ruby is a fairly high level wrapper for building voice applications. I believe some exist for PHP, but I know nothing about them. Thanks! -Ken Good luck! -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com [5] www.asterisk.org [6] Links: -- [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Command+Reference [2] http://shop.oreilly.com/product/9780596517342.do [3] http://ofps.oreilly.com/titles/9780596517342/asterisk-AMI.html [4] http://ofps.oreilly.com/titles/9780596517342/AGI.html [5] http://www.digium.com/ [6] http://www.asterisk.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inexpensive SIP Polycom conference phone?
Hey, all. It seems that Polycom has a bunch of offerings for conference phones, and I'm just wondering which are the less-expensive alternatives; what with their marketing, etc., it's not always obvious which is which. Thanks, -Ken P.S. If anyone's had really good experience with another vendor's (relatively inexpensive) conference phone, I'd also be interested in hearing about that. -- This mail was scanned by BitDefender For more information please visit http://www.bitdefender.com/links/en/frams.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Video conferencing (and SMTP server hiccups)?
Apologies for the multiple sends -- I'd been having some outbound SMTP issues, and thought the first one had fallen into the ether. Turned out, it was the upstream host that was the issue. Once kicked, lo! -Ken On Wed, 25 Jul 2012 14:24:50 -0400 Ken D'Ambrosio k...@jots.org wrote Hi, all. I'm 99% sure that Asterisk technically *supports* videoconferencing -- at least, as a conduit -- but are there products out there that leverage that? I've been tasked with bringing videoconferencing internal to my company, and had been coming up empty looking for standalone solutions, when I suddenly realized that my favorite PBX software might be able to help out. Thanks much for any pointers you might be able to give me, -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wireless SIP phone with caller announce?
I know that I could jerry-rig something that would get me caller announce from my Asterisk box, itself, but what I'd really like is a phone that does it like my Panasonics. Panasonic has a beautiful DECT/SIP series of handsets... but I guess they're aimed at the office, and jeepers, nobody wants them announcing your call *there*. (You'd think it'd be an option, off by default, but no.) Any suggestions? Thanks! -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
On Sat, February 19, 2011 4:21 pm, Ryan Wagoner wrote: On Sat, Feb 19, 2011 at 4:15 PM, Ken D'Ambrosio k...@jots.org wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? What version of dahdi do you have installed? I would try using the latest version 2.4.0. It is important to compile and install in the correct order. I usually do dahdi, libpri, asterisk, and then wanpipe. I'm running the latest of everything, except my kernel -- I went with 2.6.32.27 as being a well-maintained long-term kernel. (2.6.37 gave me grief -- too new, I guess.) I'm running -- if it makes a difference -- on an Ubuntu 8.04-4 system. I've re-installed everything, in the order you gave, to, alas, the exact same result: everything seems to initialize, install, etc., correctly, but no dahdi feature in Asterisk. Is there a module I need to load? Or... something? I'd hate to have to revert to 1.4 after all this work. Thanks! -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
You -- as usual -- hit the nail on the head; I'd actually figured it out at probably roughly the same time as you e-mailed, because I bumped into this: Asterisk Module and Build Option Selection [...] XXX chan_dahdi [...] DAHDI Telephony Depends on: dahdi(E), tonezone(E) Can use: res_smdi(M), pri(E), ss7(E), openr2(E) So, I'm using this with a Sangoma A102D; I'm not sure what the E (external?) vs. the M (module?) is about; I've compiled dahdi and tonezone -- how do I verify where the missing dependency lies? Thanks (yet again, all), -Ken On Sun, February 20, 2011 10:38 am, Tzafrir Cohen wrote: On Sat, Feb 19, 2011 at 04:15:15PM -0500, Ken D'Ambrosio wrote: Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? This may be caused by one of two things: 1. You have not built chan_dahdi.so 2. You built chan_dahdi.so, but it has failed to load (normally because of broken configuration) Try running the following from the Asterisk CLI (rasterisk): module unload chan_dahdi.so That one will likely give an error message. Ignore it. module load chan_dahdi.so What error do you get from that? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] First go at a stock 1.8 install -- where's DAHDI?
Hi, all. I've finally made the jump from 1.4 to 1.8. I've installed everything (I think), my Sangoma card initializes right... but there's no dahdi command -- not from the base, nor as a subset of the core commands. I've got my channels configured in my chan_dahdi.conf file. What am I missing, here? Thanks... -Ken -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom dial w/o Dial, while on-hook?
I've had phones before where, with the phone on-hook, it still implements the local dialplan. E.g., if I dialed 0 (on-hook), after three seconds, it would dial the operator, and have the call on speakerphone. Does Polycom allow this functionality? Clearly, not a necessary feature... but it would be a nice one. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reboot any(?) SIP Polycom -- provisioned or no.
Hey, all. I'm working on making a script to auto-provision my Polycoms. I wanted one that: - Gets the MAC by itself - Fills in the provisioning info you supplied on a web page - Creates appropriate files - Reboots the phone (which then gets provisioned) The last part was the sticking one, though. I found plenty of ways to make them reboot -- but most required an already-provisioned phone, kind of defeating my purpose. This will work with these two limitations: * The phone has default username/password * You don't care about your NAT keepalive time; I imagine most don't. (See inline comments for more info.) If that's your type of thing, enjoy! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. reboot_polycom.pl Description: / -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Soundpoint IP 430 -- discontinued.
Hey, all. I'm in the middle of a rollout, and just learned that the SoundPoint IP 430 -- my favorite mid-range phone -- has been discontinued. The heir apparent is the SoundPoint IP 450 -- for a low, low, low $130 more/handset. AND it doesn't look as nice. Ouch. Does anyone have any recommendations -- Polycom or otherwise -- for a good-quality, mid-range, two-line SIP phone (with good speakerphone) for ~$150/ea.? I realize that there are still some 430's to be had, but they won't be around forever, and now might be the right time for me to be moving forward. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ADA: DOA?
On Thu, October 7, 2010 1:02 pm, Danny Nicholas wrote: FWIW, open source is only truly dead when you can't find anywhere to download the source. I *totally* agree... if you can find me the source. I have, at this moment, at least, no reason to believe ADA is OSS -- indeed, looking at it, I see no mention of the GPL (or, for that matter, any other license), which in-and-of itself would be in direct violation of the GPL. So I'm thinking it's closed, closed, closed. Crying shame. If I'm wrong, and I hope I am, please let me know. In the meantime, if anyone gets it working -- correctly -- under 64-bit Windows, please do let me know. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ADA: DOA?
Hey, all. While ADA can still be downloaded, that's about all that I see. No development, no recent mention, and -- perhaps worst of all -- it appears not to work properly under 64-bit systems. So, assuming Digium's abandoned it, are there any suggestions of alternatives? Right now, I'm replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly fat client they have; if there's something slender -- roughly in the same line as ADA -- I'd be very interested, even if it's not free. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice-like feature.
I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice-like feature.
Want to thank everyone who mailed; a couple of your ideas got me going down certain paths, and found the answer here: http://www.voip-info.org/wiki/view/Asterisk+tips+findme Again, thanks! -Ken original message - I'd *really* like to be able to have a call ring three different cell phones; then, if someone answers, they have to somehow acknowledge the call for it to be directed to them. That way, if one of the phones is off, or out of range, it doesn't go straight to that phone's voicemail. Asterisk 1.4 -- though I could probably upgrade. Suggestions on how to make this happen? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Soft phones.
Hey, all. I'm looking -- if possible -- for a decent, multi-platform soft-phone. Specifically, Linux and Windows; that way, I'll go through the same issues my end users do. I've noticed a couple (e.g., minisip, which seems abandoned, and sip-communicator, which, honestly, is probably a great IM client, but has a confusing interface for actual phone calls). So I'm wondering if anyone has any favorites. Failing multi-platform, I'll stick with Twinkle on Linux, and gladly take suggestions for Windows -- OSS if possible, but payware is acceptable. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Multiple sip.conf files?
Hey, all. I'm trying to do some fun with auto-provisioning of Polycom phones, and one thing that would make life easier for me would be if I could have a per-phone sip.conf file. If not, no biggie -- but if there's a way to do an include (as per extensions.conf) or something, that would be great. I've gone through docs, and an older version of Asterisk: the Future of Telephony implied there was such a feature, but I've seen no mention elsewhere (including, alas, a newer version of the same book). So: can I? Or is it time to just sit down and parse the sip.conf file? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom firmware: split vs. combined
http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html Howdy, all. What's the difference between split and combined firmware, which can be seen at the above link? I've googled to no avail, I'm afraid. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVPN/SNOM 820: a review.
Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as far as I could tell, was that their firmware just doesn't seem that stable, and is sometimes hard to get to. - I managed to corrupt the firmware twice; fortunately, instead of bricking the phone, there's a fairly easy-to-use rescue mode. - Google was *not* your friend to find the URL to current firmware (for non-beta, it's http://wiki.snom.com/Firmware ; for beta, it's http://wiki.snom.com/Firmware/V8/Beta ) - There's a (non-standard) VPN release of firmware that has to be installed to get OpenVPN going. - Also got WLAN going; note that, apparently (and to my surprise), it appears that WPA keys are case-sensitive, and the phones default to uppercase. Beware. Also, you have to buy a ~$40 USB stick to get it going, but that sounds more awkward than it is: the phone has a nicely-recessed cavity on the bottom where it plugs in. Next, if you aren't familiar with OpenVPN, I *do not* recommend having the phone as your first client. Set up a Linux or Windows client, first, to get the hang of it. Then move on to the phone. For example, one of my firmware corruptions occurred when I named a file client.conf (.conf being the usual Linux-based OpenVPN configuration file extension), instead of client.cnf. Had to reflash. Bottom line: the phone actually works quite nicely. Provisioning for a one-off is a pain, but SNOM seems to have the hooks in place to make larger rollouts quite easy. OpenVPN works like a champ, but should be handled with care for those who don't have experience with it. The speakerphone quality is quite nice, and there are lots of nifty features the SNOM offers that I haven't seen on other phones -- for example, netcat is used for debugging OpenVPN, and a SIP log is truly nifty. One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BRI vs. PRI?
Hey, all. I love having a PRI to play with -- lets me do all sorts of things with DIDs, fax-to-e-mail, etc. But for a small shop, a T1 is pretty pricey. Is there any reason that a BRI can't do exactly the same stuff, but on 2B+D instead of 23B+D? Enquiring minds, etc. -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN/SNOM 820: a review.
On Thu, February 18, 2010 3:56 pm, Alex Samad wrote: On Thu, Feb 18, 2010 at 03:05:14PM -0500, Ken D'Ambrosio wrote: Hey, all. Got an SNOM 820 in the other day to kick the tires. As with many phones, provisioning it was a bit of a PITA. The biggest problem, as Thanks for the review, I was wondering if snom's mass deployment tools they have used in their other phones work with the 820's and openvpn ? Based on *inference*, I believe the tools do work, as that was the implication in things that I read. That being said, since this was a one-off, I didn't try any of the deployment tools, and could well be mistaken. -Ken [snip] One-line summary: recommended, but be prepared to spend some time getting the first one going if some of the more esoteric features (VPN, WLAN) are used. -Ken -- You're free. And freedom is beautiful. And, you know, it'll take time to restore chaos and order -- order out of chaos. But we will. - George W. Bush 04/13/2003 Washington, DC -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVPN on phones?
It's just come to my attention that newer phones from both Snom and Grandstream support OpenVPN. Is this a new trend or something? Since OpenVPN, in one swell foop, deals with both NAT issues and securing communications, I'd be very interested in hearing if other phone vendors were embracing OpenVPN as well. (Other VPN solutions are all well and good, but I really like the flexibility that OpenVPN offers.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVPN on phones?
On Thu, February 4, 2010 3:19 pm, Olle E. Johansson wrote: Anyway - is there someone out there that know the behaviour of OpenVPN in regards of retransmits and such? A VPN that retransmits will at some point hurt you if you transmit media over it, especially if you scale it up. OpenVPN allows for either TCP or UDP (and defaults to UDP) as the transport. I see no reason UDP-over-UDP would do retransmits, so I think you'd be in good shape. (Always willing to be proven wrong by someone with more OpenVPN savvy than I have, but it seems right to me.) -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linux-based hard phones?
Just wondering if there are any Linux-based hard phones out there -- if so, it'd be neat to see if I couldn't take advantage of the underlying OS. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GUI for hunt groups?
Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming extension not working.
Hi, all. I'm probably doing Something Dumb(tm), so please feel free to point out whatever I'm missing, no matter how stupid. Anyway, I've got IAX set up to Vitelity. When I try to call my DID, I get: Rejected connect attempt from 64.2.142.19, who was trying to reach '6031234567@' This leads me to my first question -- why doesn't it show a context? (My second is, what's wrong with the snippets, below?): iax.conf: [vitelity] context=vitelity register = username:passw...@inbound6.vitelity.net extensions.conf: [vitelity] ; Figured I'd try both things usually used to answer... exten = 6034713217,1,Answer exten = s,1,Answer [...] [default] include = vitelity Thanks... -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Sheeva wall wart.
Hey, all. I'm seriously thinking about doing the VoIP thing at home. The perfect platform seemed to be the Sheeva wall wart (http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp). It's a cute little doohicky with USB, SD-card, Ethernet, and runs on an ARM CPU. I'd like to avoid SIP to my provider, just 'cause it's always such a drag going through NAT and so forth. So I'd like to do IAX -- presumably a trunk (no?). Unfortunately, the Asterisk install in the Sheeva's Ubuntu distro doesn't have the IAX timing device. So I compiled Asterisk myself, and lo! Ran into the same problem the package maintainer probably did -- dahdi won't compile: dahdi-base.c:1396: error: invalid use of undefined type 'struct module' (And lots more errors of that ilk.) So: 1) Should I give up on IAX? 2) Do I need trunking? (I assume so, but...) 3) Any idea what that error's about? I, alas, am not a coder by trade. Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Receptionist GUI?
Hey, all. Just wondering if there's a GUI out there -- preferably OSS, but I'll take what-have-you -- that a) can run on an Ubuntu/Debian box, and b) allows a receptionist to see what calls are in-process, and forward calls from their phone to somewhere else. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP/WiFi handsets?
Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent price:quality ratio, of possible. Keep seeing handsets for Vonage, etc., in Best Buy and the like, but I imagine it's locked to Vonage, and can't be re-appropriated. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] * as VM for legacy PBX?
Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm wrong, a six pack of their favorite $BEVERAGE will magically appear at their door. Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * as VM for legacy PBX?
Make a call to VM (has to go out on the port you have the handset plugged into), answer it and listen. If you hear a bunch of DTMF then you are golden. Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] * as VM for legacy PBX?
Wow. Thanks for all the replies! Something just occurred to me, though: which side would be FXO, and which side would be FXS? The PBX? Or the Asterisk/VM side? Thanks again for all the info! -Ken On Wed, July 1, 2009 3:36 pm, Jared Smith wrote: On Wed, 2009-07-01 at 13:05 -0400, Ken D'Ambrosio wrote: Sounds like good stuff, but my most substantial concerns involved things like MWI: is asterisk able to push that back to the PBX? Does your existing PBX use SMDI to interface with your current voicemail system? If so, recent versions of Asterisk (1.6.0 and later, if I recall) support SMDI. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CID when using WaitExten?
Hi, all. My autoattendant looks like this: exten = s,1,Answer() exten = s,n,Background(corporate-greeting) exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds exten = s,n,WaitExten(30) When the call gets forwarded to the destination extension, however, there's no caller ID (instead, calls are from Asterisk). What am I doing wrong? Thanks! -Ken P.S. Apologies if this is a duplicate; sent originally from an account the Asterisk mailing doesn't/shouldn't know about. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom MWI.
Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go a bit over the top. The blinking LED is enough for me; how do I disable the stuttered dialtone and the audible warble? I've looked through the config files, but there are a HELL of a lot of options, and I haven't been able to find those particular ones yet. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream speakerphone?
Idle curiosity: I like the look and feel of the Grandstreams, but it's been my experience that the speakerphones suck (esp. when compared to the pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; have any of their newer models changed that? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Outlook integration?
Hey, all. I was just wondering if there were any tools/utilities/what-have-you out there that would allow a user to click on a contact in Outlook, and have their phone dial it? (Or, I guess, have Asterisk dial both their phone and the destination number, and put the two into a conference.) Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Compiling to use IMAP: how?
Hey, all. I was going through a make configure on my Asterisk 1.4.23 Ubuntu box, and noticed something I'd forgotten: Asterisk now supports IMAP_STORAGE. However, when I highlight it, it tells me that there's an unmet dependency, presumably for imap_tk. I've apt-get installed everything I can think of that might be pertinent after some Googling (e.g., uw-imapd; uw-mailutils), and nothing's changed. So: what/how do I need to install to meet this dependency? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to generate core dump?
Asterisk segfaulted on me the other day; how do I tell it to generate a core file so -- if it happens again -- I can attempt to debug? I looked in the obvious places in make menuconfig and didn't see anything appropriate. Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WiFi SIP phone w/VPN?
Hi, all. My subject line says it all: is there a WiFi SIP phone with VPN abilities? Failing that, a WiFi phone that runs Linux? I already know one phone that does meet my requirements -- the iPhone. The new software comes with a Cisco VPN client, and a SIP client can be had from third-party vendors for jailbroken phones. And, while I'm not averse to the idea, a) it ain't cheap, and b) it's a bit hack. I've googled my heart out, but haven't found anything else that (I'm sure) meets all three requirements. Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rxfax.c: Channel T30 DONE 0 -- incommplete fax reception.
Hi, all. I'm getting a lot of [Feb 3 13:56:36] WARNING[3721] /usr/src/asterisk/spandsp/agx-ast-addons/app_rxfax.c: Channel T30 DONE 0. in my log file, and incomplete fax reception. Any idea what might be going on? I've googled a fair bit, but haven't seen anything leap out at me. Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dumb question: retrieve values from OS-level commands?
Hi, all. I want to execute a script, and return the value of said (Python) script to the dialplan. I thought something like exten = 1,1,Set(MyWorkingDir=System(/bin/pwd)) might work, but apparently not. I also looked into AGI stuff, but that doesn't quite seem to be the right approach. Surely there's *some* way to do this... Any suggestions? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] extensions.conf -- what to do when command throws errors?
Hi, all. I've got app_rxfax going and nicely receiving a fax, which I
then throw to a script, and have it convert it to a PDF and mail it.
Works great... a lot of the time. But a fair bit of the time, rxfax
throws errors, and extensions.conf seems never to invoke my script. Here
are the pertinent lines:
exten = _6403,n,rxfax(${FAXFILE})
exten = _6403,n,System(/usr/local/bin/fax-sender.py ${FAXFILE})
Now the problem here is that the .TIF file is received just fine, so,
errors or no, I'd like to get to the script. Instead, I get this:
...
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] /tmp/spandsp/agx-ast-addons/trunk/app_rxfax.c: Channel T30 DONE 0.
[5410] logger.c: == Auto fallthrough, channel 'Zap/4-1' status is 'UNKNOWN'
[5410] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/4-1
[5410] chan_zap.c: Not yet hungup... Calling hangup once with icause, and
clearing call
[5410] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/4-1
[5410] logger.c: -- Hungup 'Zap/4-1'
...
In an ideal world, getting rid of the Channel T30 DONE 0 errors would
be great, but I'll take run-the-script-when-it's-done-regardless, instead.
Note, however, that I can't just call it from extension i, because I need
to pass it information, and don't want it executing on errant voice calls.
Suggestions?
Thanks!
-Ken
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[asterisk-users] Playing MP3s...
For no reason other than it would be cool, I'd like to be able to dial an extension and have it play a random MP3. Since, however, MP3s are kinda-sorta weird due to patents, I'm not sure what the right approach for this is. Any pointers on how to go about this? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] app_rxfax and app_txfax with Ubuntu?
Hi, all. I just tried to fire up app_txfax and app_rxfax, only to find that I can't seem to compile them. The problem appears to be that my libtiff library is wrong. Only problem is that, according to the README, I need libtiff =3.8 and 4.0, which is all well and good... except that there is no 3.x libtiff anywhere in the Ubuntu repository (that I can find, at any rate). So just uninstall and then install from source, right? Well, not so much: many, many different applications depend on the libtiff library, and I'd have to use --force, and... well, the whole reason I'm using Ubuntu/Debian is to avoid library/package hell. That won't cut it. Any suggestions? Oh, and, for the record, I'd prefer not to have to install Asterisk 1.6, which would appear to solve all this, just because of all the configuration I've already done. But if that's the answer, then I'll do it. -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] app_rxfax and app_txfax with Ubuntu?
What version of spandsp do you use? Based on the fact that you asked that question, I suddenly got suspicious that, despite his warnings, it might have worked for you with libtiff-4. So I went and re-tried (using spandsp 0.0.4-pre16), and it failed *differently*. So then I got suspicious that my previous spandsp install had left files lying about. I purged as best I could, re-installed 0.0.4-pre16, and it compiled fine. Now to see how well it actually works! What version of Debian (or is it Ubuntu?) For the record, it's Ubuntu Hardy. Thanks for the help! -Ken -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting up to reveive faxes.
Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] International calls/pridialplan from a legacy PBX.
Hi, all. This e-mail is a follow-up to an exchange I had several weeks ago. I've got an Asterisk box with a dual-span T1 card. I want to place it between the PSTN and my company's legacy PBX. I actually did do that, but international calls from the legacy PBX were having the 011 stripped off *AT* the PBX -- and someone pointed out that the PBX was probably using the Asterisk equivalent of pridialplan. Which makes sense. But as far as I can tell, pridialplan is used to *signal* the PSTN -- but there doesn't appear to be any way to detect it, in Asterisk, from another switch. Is that true? Because, if I can't detect it, I have no other way of determining whether or not a call from the legacy PBX is international or not, and pretty much puts the kibosh on my Asterisk plans. OTOH, if there is a way to detect it, I'm home free. Suggestions greatly appreciated! Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PoE switch recommendations?
Hey, all. We're rolling out VoIP, and I'm wondering about PoE recommendations, as we're going to have to replace our current network equipment. My first inclination would be to just plunk down the cash and do a Cisco system, but I'm relatively certain that would get shot down by finance. Any recommendations for a couple-hundred-port solution with VLANs, PoE, and QoS? Don't care much if it's in a single chassis or not, so long as it has Gbit uplinks. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bizarre international call problem.
the provider may be tagging it on. have you checked pridialplan, or prilocaldialplan settings and playing around with that in zapata.conf ? Oooh. That makes sense. I've poked around, but don't really see much documentation on this. 'Cause going outbound is easy, but how do I check to see if the inbound (from my legacy PBX) has tagged a given call as international? Thanks, again! -Ken -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do | | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bizarre international call problem.
You have handsets connected to your proprietary PBX. Most domestic things you dial on your proprietary PBX handsets get passed directly through to your asterisk box without getting mangled by your proprietary PBX. International calls that are prefixed by 011 are getting mangled by your proprietary PBX. Are you already getting to what I'm going to suggest? Modify your proprietary PBX to not mangle your international calls. Well, I really like that idea, but there's one small problem: outbound calls work just fine when the Asterisk system is removed from the equation. I'm now leaning slightly toward there being T1 funkiness between the PoS and the Asterisk box... but without a T1 protocol analyzer, it's kinda hard to be sure. Hopefully, I can get a friend over (with his) to help out. I guess -- maybe -- they could be playing games, and having the PSTN assume that calls sent out over channel X are international, but that's now sounding super-duper improbable. Thanks... -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bizarre international call problem.
Hi, all. We've got a PoS legacy PBX at my company that doesn't have call
accounting. I figured, Hey, why not stick a dual-span T1 Asterisk-based
system in the middle? Then, I just passively pass in-bound calls to the
PBX, and outbound calls to the PSTN. I can then have Asterisk do all the
call accounting, and everything should Just Work. Right?
Well, not so much.
My outbound dialing rule was incredibly complex:
exten = _X.,1,Dial(${PASSTHROUGHTRUNK}/${EXTEN})
And everything seemed to be working ducky, until I went to call Germany
and got -- a local cell phone number. Needless to say, this puzzled me
greatly. A quick look at my log, though, showed that all calls dialed
with 011 were being submitted from the PBX to the Asterisk box without
the 011. (Ironically, if I dial the number with 011011 in front, it
goes through fine.)
So I'm confused: any ideas on how this worked when the PBX was hooked
straight to the PSTN? Is there some SS7 signal or something that says,
This is an international call, when the number has no 011 preface? I'd
hate to have to revert, but I will if need be... *sigh*
Thanks for any insights. I'm totally flummoxed.
-Ken
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[asterisk-users] extensions.conf programming?
Hey, all. I haven't really gotten deep into Asterisk since 1.0.x, and I'm afraid I've forgotten a fair bit. One big thing that I've forgotten is the syntax, etc., for extensions.conf. Where do I find that? I'm looking for stuff about commands, syntax for commands, variables, etc. Is there a handy-dandy manpage, webpage, or what-have-you that I'm missing? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Selectively disable echo cancellation?
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm currently passing through some of my in-bound calls to a legacy PBX (which I hope to eventually replace). That being said, until I do, I'd like to kill echo cancellation for the passed-through calls -- I don't want to mess with their fax reception. Any idea how to do this? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk end-user GUI?
I badly want to roll out Asterisk at my job. Unfortunately, my boss is dazzled by shiny objects. We had a vendor in today who showed us their system which, honestly, didn't suck -- but boy, is it going to be expensive! One major component of the eye candy was an end-user interface that allowed the user to initiate calls to a contact list, check for presence, create conferences, etc. Is there anything like that, aimed at end-users (as opposed to admins) for Asterisk? I'd even be willing to go with proprietary; I just don't want a wholly-proprietary, hobbled, licensed-to-Heck-and-back system, which is where it looks like my boss is leaning. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk end-user GUI?
[Sorry if this is a duplicate; originally sent from an address the list doesn't know.] Wow. Okay, Druid has my attention; I'll definitely be kicking the tires. That being said, though, I do have a quick question (that I always have about GUIs): First, I assume that Druid is based on Asterisk; is this true? Second, is it possible to make system modifications w/o using the GUI? I love GUIs, but sometimes there's just nothing as cool as a quick Perl script. Thanks! -Ken Ken, You might want to check out our free Druid Open source unified communications project. It is not proprietary and has open source soap API for third party applications. http://www.voiceroute.org We have mobile integration with blackberry iphone that no vendors open source or otherwise has. Our Druid SOAP API that powers this integration is free. Check out our youtube oscon presentation on Druid SOAP API http://www.youtube.com/user/voiceroute Ming On 8/7/08, Ken D'Ambrosio [EMAIL PROTECTED] wrote: I badly want to roll out Asterisk at my job. Unfortunately, my boss is dazzled by shiny objects. We had a vendor in today who showed us their system which, honestly, didn't suck -- but boy, is it going to be expensive! One major component of the eye candy was an end-user interface that allowed the user to initiate calls to a contact list, check for presence, create conferences, etc. Is there anything like that, aimed at end-users (as opposed to admins) for Asterisk? I'd even be willing to go with proprietary; I just don't want a wholly-proprietary, hobbled, licensed-to-Heck-and-back system, which is where it looks like my boss is leaning. Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sent from Gmail for mobile | mobile.google.com Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Meet us at LinuxWorld 2008, 4-7 Aug 2008, Moscone Center, San Francisco, Booth 1626 http://druidlinuxworld.eventbrite.com Meet us at WEB 2.0 EXPO, 16-19 Sept 2008, Javits Center, NYC, Booth #17 http://druidweb20.eventbrite.com See Voiceroute OSCON 2008 Druid project presentation on youtube http://www.youtube.com/watch?v=2gfIAXm5vTc http://www.youtube.com/watch?v=dkm6P4O0oac ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad recorded audio quality (upgrade).
Hi, all. I'm doing an upgrade from an [EMAIL PROTECTED] (Asterisk 1.x) system to stock Asterisk 1.4. Everything's working great, except that all the prompts (both stock system prompts on the new system and people's old recorded VM prompts) sound HORRIBLE. Call quality is great, both internal and external. Any idea as to what might have happened? Could I have brought over a config that's not valid for this setup? Thanks! -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] *#%! Polycom...
I used to do lots of Asterisk, but got an offer I couldn't refuse, and went SysAdmin. Well, now I'm trying to bring Asterisk in-house, and want to set up a test system. One thing I'd really like to get my hands on is recent firmware, etc., for SoundPoint IP 430's. Freedomphones.net, my old source, seems to have been kaput about as long as I've been a sysadmin; are there any other sources out there? (And, yeah, if anyone wants to e-mail them to me directly, I won't say no.) Thanks much, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for a cheap SIP termination point.
Hi, all. I'm trying to do some rudimentary testing of an Asterisk system, but, for various reasons, I have to do this covertly, which means I'm paying out-of-pocket. So I'm looking for somewhere that will do *cheap* SIP and/or IAX termination, preferably with at least two simultaneous calls, and one DID.Any suggestions? Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP 330 w/VLAN?
Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Emagen (a Telrad VM solution) -- any way to replace with *?
Hi, all. I've got a PoS Emagen VM system tied in with our Telrad PBX. I hate 'em both, but I'm stuck with the Telrad for the time being. That being said, does anyone know of a way to replace the VM solution with Asterisk? I'd -love- to get an Asterisk box in the loop, here. Thanks, -Ken ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real API for Perl?
Hi, all. I've used the perl/AGI interface, and... well, I found it kind of hokey. Granted, this was in 1.2 days -- perhaps things have changed. Regardless, I guess I have two questions: 1) Has the Perl/AGI binding improved since then? 2) Is there any chance of a real API for Perl? Thanks much! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Your favorite Asterisk application.
Hi, all. I've done some Asterisk recelling, but recently got roped into a Sr. SysAdmin position. Our PBX is c. 1823, and -- well, as pretty much all circuit-based systems do, it sucks. It sucks to administer, moves suck... you know the drill. So, I'd love change to an Asterisk system. My boss, who loves to spend money for no particular reason, wants to go proprietary, though. So I'm going to have to try to sell him. I figured one place to start would be some of the really cool applications that Asterisk has that -- generally, at least -- don't require licensing. Some of my favorites are follow-me, meetme, voicemail-to-e-mail and fax-to-e-mail. What are some of your favorite features/applications, be ith native or third-party? Thanks, -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] German SIP and/or IAX providers?
Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. However, I'd need a few things: - Reliability. Can't have my branch office's DID's just going down. A company with a proven track record would be very, very good. - English. I speak great English, decent Spanish... and zilch German. So, as provincial as it might make me, I need a company I could talk to if the chips are down. And that's about -it-. I'm even willing to pay a reasonable premium, so long as it gets me a VoIP provider with the above restrictions. Any suggestions? Thanks much! -Ken ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Useful GUI? [Was: Why does everyone seem to dislike *now?]
On Mon, September 17, 2007 7:21 pm, Matt Riddell wrote: In the past, you could help someone sort a problem, only for the config files to be overwritten the next time the user did something in the GUI. Are there any Asterisk GUIs out there that actually parse the data files, themselves, instead of having some sort of metadata middle-man, which leads to said overwriting? I mean, I, personally, love the CLI -- always have been a fast typist -- but I also know the CLI would scare the living bejeepers out of my boss if/when I try to push hard on an Asterisk solution. What I'd prefer is: - The chance to do CLI stuff as I see fit, BUT - the ability to let users -- even administrative users -- use a GUI, without messing up my beautiful config files. Is this a pipe dream, or is there a GUI out there that might actually do the job? Thanks, -Ken ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail in 1.4?
I got dragged away from Asterisk (somebody made me an offer I couldn't refuse for system administration), but I'm thinking about seeing if I can't get it deployed at my new employer. Regardless, there are two things about older voicemail that used to annoy me: - Dial by name. Has anyone made it so it can be first or last? - Jump to voicemail; you used to have to actually dial the voicemail, whereas most voicemail systems allow you to go to your mailbox when you hear your voice prompt. Any chance this has been rectified? Thanks, -Ken ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Volume (gain?) on VoIP-only system.
Hi, all. I've got a customer who's complaining of low volume, especially for conference calls. If this were a Zap system, I'd just bump up txgain in their zaptel.conf file... but it isn't. Should I crank the volume of the phones (they're Polycoms), or is there some other, more graceful, system-wide setting I could use to increase gain? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Polycom: warble on registration?
Hi, all. I just upgraded my sip.cfg for my Polycoms, and that damn warble on registration(? -- maybe it's on acquiring an IP?) has started again. I still have the old sip.cfg, but can't figure out which option it is. Any help? Thanks! -Ken D'Ambrosio -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Disappearing voicemail?
Hi, all. Today, our receptionist got an e-mail saying she had a 55-second voicemail... but the attachment was 0 bytes. Turns out, so was the message when accessed via the phone. A quick purview of the logs turned up this: VERBOSE[14836] logger.c: -- Playing 'vm-savefolder' (language 'en') DEBUG[14836] channel.c: Scheduling timer at 0 sample intervals WARNING[14836] file.c: Failed to write frame DEBUG[14836] channel.c: Scheduling timer at 0 sample intervals VERBOSE[14836] logger.c: -- Playing 'vm-mailboxfull' (language 'en') [Timestamps removed for readability; the latter four lines all occurred within one second of each other; the first line happened two seconds prior.] Anyway, I went through the logs, and noticed that this had happened four previous times. Not enough, over the course of the four months involved, to be overly concerned, but any time voicemail simply goes away, it's probably worthy of note. Is this a known issue? Is there some way around it? I've Googled, to no avail. Thanks! -Ken ___ --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] IAX phones?
Just wondering if there are any IAX phones worthy of the name phone out there -- looking for hard phones, but I suppose a Linux-based softphone wouldn't, you know, hurt. ;-) Thanks! -Ken ___ --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] does /var/run/asterisk.ctl exist? -- but Asterisk *is* running.
I've set up a bunch of plain-jane Asterisk systems, but had heard good things about the more recent incarnations of [EMAIL PROTECTED] errr, Trixbox. So I installed it, and fired it up, and it works fine. Until I try to do an asterisk -r. I get the does /var/run/asterisk.ctl exist? question, which had always previously meant (to me) that Asterisk wasn't running. But it is! And there's now asterisk.ctl file in the entire /var hierarchy. Anyone have any ideas as to why that might be MIA? It's insanely annoying, not being able to fire up the console. Thanks, -Ken ___ --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] Polycom SoundPoint 2.0.1 SIP firmware?
Hi, all -- since the new 2.x firmware seems to support NAT -- and, since I'm not an authorized dealer -- I'm kind of wondering if anyone knows where I can get it. freedomphones.net/polycom/files/ only goes up to 1.6.7. If anyone can either mail it to me, or mail me a link, I'd certainly be appreciative. Thanks! -Ken ___ --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] Sipura 3000 dialplan strings.
I'm trying to set up a dialplan that dials via PSTN for: All eight-digit calls that start with 9 All 911 calls All calls that start with 424 (the local exchange) I haven't tested 911 -- for obvious reasons. I may do so after I feel more confident. I've got the starts-with-9 working fine. But the local exchange stuff isn't working, and I'm confused. Here's a snippet of my dialplan: [lots deleted]|9,:xxx :@gw0|424 :@gw0) It does dial 424 numbers, but they go straight through SIP. Any suggestions? Thanks! -Ken ___ --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] Analog-to-VoIP: blade?
I've seen analog-to-VoIP gateways such as the Audiocodes one -- which, truthfully, looks very, very nice -- but I've got several hundreds of analog phones to deal with, and I was wondering if anyone has seen something with even higher concentrations than the Audiocodes 24-ports-per-rack-unit. Thanks for any suggestions! -Ken ___ --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] No zap command?
Hi, all. I've just set up an Asterisk box -- to the best of my knowledge, no differently than any of the others that I've set up. Only one minor caveat: there's no zap command. Huh? Glancing at the startup, there's no mention of chan_zap, which I assume is partially the reason. However, I'm using -the exact same- zapata.conf, extensions.conf, and zaptel.conf from a different install, so I would imagine it would have been invoked if it were a config issue. Is there a compile-time option that we missed? [And, for the record, no zap errors whatsoever in the log. So it's not like it's trying to load chan_zap.o and failing or anything.] Any ideas would be greatly appreciated... Thanks, -Ken ___ --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] Hotels...
I have to bid on a hotel contract, but there are some things I don't know how to do -- but clearly Asterisk has been used by hotels before, so I figure someone on here must have some answers: 1) While the majority of the phones will be SIP, there will be a couple hundred analogs (due to wiring logistics); what should I terminate them into? 2) Phone activation at check-in/phone de-activation and billing at check-out. Are there GUI tools for this, or should I write my own back/front end? 3) Anything else that those familiar with hotels have bumped into that might not be obvious at the outset? Thanks! -Ken ___ --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] How to find out which line in extensions.conf?
When trying to figure out why something's not working, is there any way to have the output specify which line of extensions.conf was being executed? I mean, sure, I could pour a million NoOp()'s into it, but that's not exactly scalable, nor easy. It would be really nice if, instead, along with timestamp, it mentioned either a line number, or -- more likely -- a context/extension/priority triplet. Is there anything like that? 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] Reception softphone suggestions?
Hey, all. I've got a client who's interested in possibly using a softphone for his receptionists. While I've certainly used some softphones for single extensions, I'm not sure which one I'd suggest for a receptionist. Any favorites? Thanks, -Ken ___ --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] In-bound faxing working ~1/3 of time.
Hey, all. I've got inbound faxing going with SpanDSP's RxFax application. And, when it works, it works great. However, roughly 2/3 of the time, it fails partway through the fax, and the sending fax machine breaks contact. I'm running Asterisk 1.2.4, which works fine in al lother particulars. I'm not sure of the SpanDSP version, but it's a recent one. Oh, forgot to mention: I have a range of DIDs pointing to my 23rd B channel on my T1, which then goes into my Sangoma A104d card; I've got echo cancellation turned off (to prevent wonkiness in the fax reception; with it on, the latency screws stuff up). Any ideas? 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] Error on Polycom 501 601.
Hi, all. Every now and then, some of my users get Error on their phones. A reboot fixes it, but it's quite annoying/inconvenient. I'm running Asterisk 1.2.4, and have the following firmware, etc.: Bootrom: 2.6.2.0032 BootBlock: 2.5.0(11500_030) SIP application: 1.6.2.0041 Any ideas as to why this might be happening? Thanks! -Ken ___ --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] Polycom 601 -- programming buttons.
Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. 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] CallerID/variable setting.
Hey, all. I'm trying to set my CID such that, internally, I see a
four-digit extension (which is also handy when checking VM), but
externally, I see the full 10-digit number. So I plugged these lines into
my extensions.conf:
exten = _XXX,1,GotoIf($[ ${CALLERIDNUM} != 1625]?4:2)
exten = _XXX,2,Set(CALLERIDNUM=6031234${CALLERIDNUM:1})
exten = _XXX,3,NoOp(${CALLERIDNUM})
exten = _XXX,4,Dial(${OUTBOUNDTRUNK}/${EXTEN})
(I wanted to test against my own extension, 1625; if that worked, I
wanted to strip off the 1, and then prepend the 603-123-4 to my
remaining three digits.)
Which is all well and good -- until I actually try to use it. Then, I get:
-- Executing GotoIf(SIP/1625-f89a, 0?4:2) in new stack
-- Goto (internal,7654321,2)
-- Executing Set(SIP/1625-f89a, CALLERIDNUM=6031234625) in new stack
-- Executing NoOp(SIP/1625-f89a, 1625) in new stack
-- Executing Dial(SIP/1625-f89a, Zap/g1/7654321) in new stack
Why does my NoOp line show my 1625 extension, when CALLERIDNUM is -- as
far as I can tell -- being set to 6031234625? (I looked against the Set
command page on the Wiki, and I think I'm doing it right.)
Asterisk 1.2.3, if that matters.
Thanks,
-Ken
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Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3
Douglas Garstang wrote: Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Fascinating, captain. "The docs" to which I refer are the "Administrator Guide - SoundPoint/SoundStation IP SIP", v. 1.3.0, where, on page 13, it states: The following FTP servers have been tested with Soundpoint IP and are known to work acceptably: Linux: ProFTPd 1.2.2 through 1.2.9 rc2p, ftpd-bsd-0.3.3 (Linux port), we-ftpd 2.6.0 Windows 2000 Server: IIS 5.0, WFTPD 2.03 So, I went surfing, to find which page it's on in the latest-and-greatest doc (v. 1.6.x, found at www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,4922,00.pdf ), and guess what? They've taken the section out. I guess I don't know what the "supported" FTP servers are, any more. Perhaps they're now fully-compliant with "stock" FTP transactions, and It Just Works. Dunno. Color me confused, -Ken P.S. Note, however, that I'd tried... umm, whatever the stock FTP that comes with Debian is, and it failed miserably -- even though I could FTP in just fine from the command line, AND an Ethereal dump showed that the FTP transactions were being executed properly, but the phone wasn't responding correctly. It was only when I went with ProFTPd that things got better -- for me, at least. ;-) YMMV, etc. Doug. -Original Message- From: Ken D'Ambrosio [mailto:[EMAIL PROTECTED]] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3 HTTP's nice, but FTP does the job. Check the docs for supported FTP servers -- many of the stock Linux FTP servers will give the exact problem you discussed, below. I should know -- took me almost a week before trying proftpd, and WHAMMO, worked like a champ. -Ken On Tue, March 7, 2006 12:37 pm, William M Conlon wrote: I spent a weekend battling similar issues with 501s, using FC4/ proftpd. I finally switched from FTP to HTTP. On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote: Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated bootrom, nice. Rebooted again. Updated sip firmware. Also nice. Upon the next reboot, the wheels started falling off. The phones would not get changes I made to any of the .cfg files. Several phones would take 20 minutes or more to boot, only to display a "0x4000 config file error". What happened? I have been using various Polycom's with AstLinux (and vsftpd 2.0.3 that I include with it) for quite some time, with no problems whatsoever. Until now. I had been running bootrom 3.0.1 and various versions of the SIP image at several other sites with no problem. At this point I was still unable to accept the fact that I might not be able to run this latest bootrom. After many trial and tribulations, I finally rsync'ed (with -avr) the FTP directory from the AstLinux machine to my laptop running CentOS 4. I configured the vsftpd daemon (version 2.0.1) IDENTICALLY (with the exception of PAM and TCP wrappers) and crossed my fingers... After re-configuring the IP 501 to use my laptop, imagine my surprise when the most problematic of them booted right away without problems. Again and again, everything was fine. So now I just had to break out ethereal and see what was going on. While I have not completely finished my analysis, it appears that Polycom firmware 3.1.3 bombs out when transferring files with vsftpd 2.0.3. The symptom appears to be repeated TCP SYNs from the Polycom to the ftp daemon on port 20. The Polycom will keep retrying and increment its source port number by one every few minutes. Like I said, I need to dig into this more, but I figured I'd report what I know and see if anyone out there can fill in the holes. Here's what I did. It appears that BootRom 3.1.3 works with vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd 2.0.3) on my CentOS server and downgraded the phone to 3.0.1. I then placed 3.0.1 and SIP app 1.6.5 (which I was using the whole time, btw) on my AstLinux server running vsftpd 2.0.3. All was good. So now I am successfully running with the following: Polycom IP 501 Bootrom 3.0.1 SIP 1.6.5 AstLinux 0.3.7 vsftpd 2.0.3 I will also try to fix (or workaround) this by trying the following: upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate BootRom release between 3.0.1 and 3.1.3 (find out exactly where/when it broke) trying an even newer P
Re: [Asterisk-Users] OT: Polycom BootRom 3.1.3 and vsftpd 2.0.3
HTTP's nice, but FTP does the job. Check the docs for supported FTP servers -- many of the stock Linux FTP servers will give the exact problem you discussed, below. I should know -- took me almost a week before trying proftpd, and WHAMMO, worked like a champ. -Ken On Tue, March 7, 2006 12:37 pm, William M Conlon wrote: I spent a weekend battling similar issues with 501s, using FC4/ proftpd. I finally switched from FTP to HTTP. On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote: Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated bootrom, nice. Rebooted again. Updated sip firmware. Also nice. Upon the next reboot, the wheels started falling off. The phones would not get changes I made to any of the .cfg files. Several phones would take 20 minutes or more to boot, only to display a 0x4000 config file error. What happened? I have been using various Polycom's with AstLinux (and vsftpd 2.0.3 that I include with it) for quite some time, with no problems whatsoever. Until now. I had been running bootrom 3.0.1 and various versions of the SIP image at several other sites with no problem. At this point I was still unable to accept the fact that I might not be able to run this latest bootrom. After many trial and tribulations, I finally rsync'ed (with -avr) the FTP directory from the AstLinux machine to my laptop running CentOS 4. I configured the vsftpd daemon (version 2.0.1) IDENTICALLY (with the exception of PAM and TCP wrappers) and crossed my fingers... After re-configuring the IP 501 to use my laptop, imagine my surprise when the most problematic of them booted right away without problems. Again and again, everything was fine. So now I just had to break out ethereal and see what was going on. While I have not completely finished my analysis, it appears that Polycom firmware 3.1.3 bombs out when transferring files with vsftpd 2.0.3. The symptom appears to be repeated TCP SYNs from the Polycom to the ftp daemon on port 20. The Polycom will keep retrying and increment its source port number by one every few minutes. Like I said, I need to dig into this more, but I figured I'd report what I know and see if anyone out there can fill in the holes. Here's what I did. It appears that BootRom 3.1.3 works with vsftpd 2.0.1, so I placed bootrom 3.0.1 (which I know works with vsftpd 2.0.3) on my CentOS server and downgraded the phone to 3.0.1. I then placed 3.0.1 and SIP app 1.6.5 (which I was using the whole time, btw) on my AstLinux server running vsftpd 2.0.3. All was good. So now I am successfully running with the following: Polycom IP 501 Bootrom 3.0.1 SIP 1.6.5 AstLinux 0.3.7 vsftpd 2.0.3 I will also try to fix (or workaround) this by trying the following: upgrading AstLinux to include vsftpd 2.0.4 trying an intermediate BootRom release between 3.0.1 and 3.1.3 (find out exactly where/when it broke) trying an even newer Polycom BootRom when it becomes available upgrading the kernel in AstLinux (I doubt that's it) fiddling with iptables rules in AstLinux (iptables was loaded, but obviously 3.0.1 doesn't have a problem with it) This also might be related to the problems described here: http://forums.digium.com/viewtopic.php? p=14847sid=6e70577c37bd345cfc164a01e64e113a Any thoughts? Comments? Suggestions? P.S. - I will be updating the Polycom config files at http:// www.krisk.org/asterisk/pcom/ to reflect some new changes in this firmware release. I just need to get my phones working first :)! -- Kristian Kielhofner ___ --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 Bill William M. Conlon, P.E., Ph.D. To the Point 345 California Avenue Suite 2 Palo Alto, CA 94306 vox: 650.327.2175 (direct) fax: 650.329.8335 mobile: 650.906.9929 e-mail: mailto:[EMAIL PROTECTED] web: http://www.tothept.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
[Asterisk-Users] Initiate and monitor multiple calls?
I'd like to set up a sort-of follow-me: on a call to a given extension, I'd like to simultaneously call several different numbers, play them all a prompt upon answering, and monitor for DTMF digit 1. I know how to get Dial() to dial multiple numbers, and I know how to play prompts and monitor for digits... but I don't know how to mix it all together. Any pointers on where to start looking? Thanks! -Ken ___ --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] Polycom 501 power over ethernet
On Mon, March 6, 2006 4:19 pm, [EMAIL PROTECTED] wrote: I have installed several hundred polycom's, and I have never seen a 500/501 with a power jack. All with the inline cable, as you mention. Of course, if someone can provide photo evidence I will stand corrected. I think the confusion here is the different *ways* the 300/500/600 do PoE: 301 has a power brick, just like (say) a Grandstream. 501 has _almost_ PoE: the cable is (as noted above) in-line, but this might confuse someone differentiating with the 301. 601 has true PoE, where you've got your PoE switch, a stock Ethernet cable, and the phone -- nothing else, and no special cabling required. -Ken (purveyor of fine differentiations) PaulH - Original Message - From: The VoIP Connection [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 07, 2006 4:26 AM Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet I've seen a lot of IP501 and I've never seen one with a power jack. According to Polycom they all use the cable. Possibly it was an IP500? -Mike Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Douglas Garstang [mailto:[EMAIL PROTECTED] Sent: Monday, March 06, 2006 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet No, some IP 501's have the inline cable and some have the power jack. -Original Message- From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Sunday, March 05, 2006 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom 501 power over ethernet The IP300/301 has the power jack, the IP500/501 the inline cable. PaulH On Sun, 2006-03-05 at 20:56 -0700, Douglas Garstang wrote: Not true. Some do and some don't. Some have a place to plug a separate DC adapter, and some have the inline power, where the adapter plugs into the ethernet cable. Not sure which ones are newer, and which are older. -Original Message- From: Michael Welter [mailto:[EMAIL PROTECTED] Sent: Sun 3/5/2006 6:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Polycom 501 power over ethernet The IP501 does not have a power jack. You'll need one of the Polycom cables. William M Conlon wrote: My recollection of the marketing fluff was that we would just use our legacy network (cables) and the devices at both ends would figure out whether they were sourcing, sinking, or neither. In the case of the 501, it's the special Polycom cable, either with or without provision for an AC power adapter, that powers the phone. That's what I meant by saying the '501' itself is not compliant with 802.3af -- it needs a separate thingamajig [tech jargon :)]to be powered. Anyway I had hoped that I could just plug a CAT-5 patch cable from my RJ45 wall outlet into the phone. On Mar 5, 2006, at 5:17 PM, Michael Welter wrote: As I understand 802.3af, the phones go through a negotiation with the unit supplying the power. I don't think it's a matter of -48VDC on a particular pair. I remember a schematic from years ago--it had each of the receive pair and the transmit pair going into a transformer winding, and that winding had a center tap for PoE. This is not something that *I* am going to screw with. The IP501 telephone set is the same for both PoE and local power. With the PoE cable, the 802.3af electronics (the negotiator) is a plastic thing in the cable. For the local power, there is a plastic thingie toward the wall end of the cable, and you plug the wall wart into the plastic thingie. Notice the advanced technical jargon here With local power, there is still only one cable one the desk--the power plugs into the cable towards the wall. Except for a power interruption, this has all the advantages of PoE. William M Conlon wrote: I saw that Polycom offered a cable (not stocked anywhere), at $40 a pop for 802.3af connections. That's what made me think the phone itself is NOT 802.3af compliant. Presumably, for $40, there's more than a fuse in that special cable. On Mar 5, 2006, at 4:31 PM, Paul Hales wrote: For Polycom IP500/501's and IP300/301's you need a special polycom POE cable. When you buy Polycom phones you can usually specify POE or powerpack. PaulH On Sun, 2006-03-05 at 16:23 -0800, William M Conlon wrote: When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were powered over ethernet. Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear [EMAIL PROTECTED]) won't power them, though if I use
[Asterisk-Users] No audio on PRI.
Hi, all. I've just had my T1 re-provisioned to ISDN. Everything comes up and seems to work fine, with the minor detail that there is no audio whatsoever. So: voice prompts are played, caller ID and DID information is seen and acted on, etc., etc., etc., but at no point is any audio heard on either in-bound or out-bound calls. Here, respectively, are my zaptel.conf, zapata.conf, and the dump of pri debug span 1: http://pastebin.com/583049 http://pastebin.com/583050 http://pastebin.com/583060 Any ideas would be greatly appreciated. Note that I'm reasonably sure of the zaptel.conf and switch type. Thanks, -Ken ___ --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] Choice One PRI?
Hi, all. I've got a T1 through Choice One Communications (www.choiceonecom.com), a provider in the US northeast. I recently tried to switch to ISDN on it -- and failed miserably. I've posted my config files, and nobody's seen anything obviously wrong. Has anyone else used their ISDN T1's? If so, would you be kind enough to send me your zapata.conf and zaptel.conf files? Thanks! -Ken ___ --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] No D-channels available!
I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty badly. Seemingly everything worked -- Asterisk would see the incoming call (including CID and DID info), try to route it, and fail -- giving me a telco (not Asterisk) call failure message. My zapata.conf and zaptel.conf files are at http://pastebin.com/558349 Below's the log dump. Note that, because I was simply going from CAS to PRI, I don't believe the span definition line, itself, should have changed. If anyone has any ideas, I'd sure be interested... -- Accepting call from '1234567890' to '1630' on channel 0/1, span 1 Enabled echo cancellation on channel 1 -- Executing Goto(Zap/1-1, internal|1630|1) in new stack -- Goto (internal,1630,1) -- Executing Dial(Zap/1-1, SIP/1630|18) in new stack Setting NAT on RTP to 0 Outgoing Call for 1630 -- Called 1630 (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/1630-9ce0 is ringing Requested indication 3 on channel Zap/1-1 !! Got reject for frame 0, retransmitting frame 0 now, updating n_r! !! Got reject for frame 0, retransmitting frame 1 now, updating n_r! Echo cancellation already on == Primary D-Channel on span 1 down No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 up update_call_counter(1630) - decrement call limit counter ___ --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] Sangoma analog cards?
Michael Graves wrote: Does anyone on-list have direct experience with the new analog cards from Sangoma? I'm thinking about FXOs with echo cans. Need 2-4 ports but don't want to go through another TDM400 style experience. First impressions (of which one should always be wary): 1) I really, really like the look of the new card. From a mechanical engineering standpoint, it's a joy to be able to simply snap a module in, or remove it. Beats the TDM400 hands-down in the ME department. 2) Looks can be deceiving: I had a devil of a time trying to get my T1 and analog cards working in the same machine at the same time. Caveat: I've never done it before with any hardware, and may have been Doing Something Dumb(tm). Haven't tried getting just the analog card working by itself, since the analog was just for conference room phones, and I care far more about the T1. 3) My T1 card is a Sangoma with EC -- assuming they use the same EC on the analog as the T1 card (check -- I really don't know), I gotta say it works pretty darn well. All my echo is -gone-. -Ken Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ --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] No D-channels available!
Michael Collins wrote: Ken, The zaptel.conf looks good as far as I can tell. The only question I have is on the Zapata.conf - do you know for sure that the switchtype is supposed to be national? Just curious. My telco's are all set for 4ess/5ess or dms100. Yah, it is national -- not that that kept me from trying everything else for the hell of it. (I tried, in order, national, ni1, 4ess and 5ess. None worked, but national -- which is what they told me the switch was, to start with -- came by far the closest.) Second, can you do a debug pri span 1 from the CLI and see what the output is? I'm assuming that there's at least *some* communication on the D-channel since you're able to get inbound calls. I'm just curious to know what's happening on the D channel when you are dialing out. I can't. It was a one-shot deal, as (because of the phone company) I can only get the T1 turned to ISDN during work hours, which means that my company's lines are down while I'm trying to switch over. I realized about 1 minute after I told them to revert that I should have gotten a debug pri dump, but it was too late. -Ken -MC -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Thursday, February 16, 2006 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] No D-channels available! I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty badly. Seemingly everything worked -- Asterisk would see the incoming call (including CID and DID info), try to route it, and fail -- giving me a telco (not Asterisk) call failure message. My zapata.conf and zaptel.conf files are at http://pastebin.com/558349 Below's the log dump. Note that, because I was simply going from CAS to PRI, I don't believe the span definition line, itself, should have changed. If anyone has any ideas, I'd sure be interested... -- Accepting call from '1234567890' to '1630' on channel 0/1, span 1 Enabled echo cancellation on channel 1 -- Executing Goto(Zap/1-1, internal|1630|1) in new stack -- Goto (internal,1630,1) -- Executing Dial(Zap/1-1, SIP/1630|18) in new stack Setting NAT on RTP to 0 Outgoing Call for 1630 -- Called 1630 (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/1630-9ce0 is ringing Requested indication 3 on channel Zap/1-1 !! Got reject for frame 0, retransmitting frame 0 now, updating n_r! !! Got reject for frame 0, retransmitting frame 1 now, updating n_r! Echo cancellation already on == Primary D-Channel on span 1 down No D-channels available! Using Primary channel 24 as D-channel anyway! == Primary D-Channel on span 1 up update_call_counter(1630) - decrement call limit counter ___ --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 ___ --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
