GXP 2000 firmware - warning
Hi all, anyone who needs to upgrade their firmware should NOT download from www.grandstreamsucks.com this firmware will permanently cripple your phone. Henry
surprising google purchase
http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP...
Re: [on-asterisk] surprising google purchase
Sarcasm What innovation! /Sarcasm Not only is it a commodity feature on many systems like Asterisk but it's already been commercialized for years. (IE Bell's Prime Line). 50M isn't too much money, maybe they've got infrastructure in place and a staff to manage it. Just having that out of the way would get you to market a year quicker. DD On 7/4/07, Simon P. Ditner [EMAIL PROTECTED] wrote: http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
RE: [on-asterisk] Calls not connecting on PRI span
David:
What's you zaptel.conf say? Are both groups set to pri_net or pri_cpe
signalling? You'll need that for a crossover cable...
Thanks,
Mark.
Please note: Effective April 1, my e-mail address is
[EMAIL PROTECTED]
-Original Message-
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 03, 2007 1:57 PM
To: asterisk@uc.org
Subject: [on-asterisk] Calls not connecting on PRI span
Here is an interesting support problem.
New A102D card. Span 0/1 tied with a cross over cable (to test).
Dial an extension out one span
exten = _30X,1,Dial(Zap/G0/${EXTEN},45)
Should dial out on last channel of group 0 and should arrive on last
channel of group 1 (other span)
[from-internal]
; inbound from (group 1) Nortel defaults to pass-thru to PSTN
; pstn on trunk group 0, descending order
exten = _.,1,MusiconHold()
I should hear music. However I get no connection ... Last message is
proceeding below, no ringing yet the channel (47-1) won't hangup.
It's like it doesn't know to connect the two legs of the call!!
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/7046-1bebd770,
Zap/G0/300|45) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/300
-- Accepting call from '7046' to '300' on channel 0/23, span 2
-- Executing [EMAIL PROTECTED]:1] MusicOnHold(Zap/47-1, ) in
new stack
-- Started music on hold, class 'default', on Zap/47-1
-- Zap/23-1 is proceeding passing it to SIP/7046-1bebd770
topbx*CLI
Asterisk 1.4.5
Zaptel 1.4.3
Sangoma Wanpipe 2.3.4-10
Any ideas??
dbc.
--
David Cook
-
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Re: [on-asterisk] surprising google purchase
Cash burning holes in their pockets.MUST SPEND !!! - Original Message - From: Simon P. Ditner [EMAIL PROTECTED] To: asterisk@uc.org Sent: Wednesday, July 04, 2007 9:48 AM Subject: [on-asterisk] surprising google purchase http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
RE: [on-asterisk] surprising google purchase
Wasn't there someone in Ottawa who had a similar business? They would assign one phone number to you and it would be forwarded to various phones. I seem to recall thinking that it was quite cool so maybe Google is hoping for the same. (note that this service only applies to USA ) Though you would still need a provider whether it's a cell service, land line etc. to forward this number and voice mail to. Maybe they'll tie it in with their gmail for a total communication tool. ?? From: Dave Donovan [mailto:[EMAIL PROTECTED] Sent: July 4, 2007 9:12 AM To: asterisk@uc.org Subject: Re: [on-asterisk] surprising google purchase Sarcasm What innovation! /Sarcasm Not only is it a commodity feature on many systems like Asterisk but it's already been commercialized for years. (IE Bell's Prime Line). 50M isn't too much money, maybe they've got infrastructure in place and a staff to manage it. Just having that out of the way would get you to market a year quicker. DD On 7/4/07, Simon P. Ditner [EMAIL PROTECTED] wrote: http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] surprising google purchase
Are you thinking of Iotum's presence management suite? I think they were tightly integrating things with Exchange, but have now re-focused on the Blackberry market. On 7/4/07, Mike Dancy [EMAIL PROTECTED] wrote: Wasn't there someone in Ottawa who had a similar business? They would assign one phone number to you and it would be forwarded to various phones. I seem to recall thinking that it was quite cool so maybe Google is hoping for the same. (note that this service only applies to USA ) Though you would still need a provider whether it's a cell service, land line etc. to forward this number and voice mail to. Maybe they'll tie it in with their gmail for a total communication tool. ?? From: Dave Donovan [mailto:[EMAIL PROTECTED] Sent: July 4, 2007 9:12 AM To: asterisk@uc.org Subject: Re: [on-asterisk] surprising google purchase Sarcasm What innovation! /Sarcasm Not only is it a commodity feature on many systems like Asterisk but it's already been commercialized for years. (IE Bell's Prime Line). 50M isn't too much money, maybe they've got infrastructure in place and a staff to manage it. Just having that out of the way would get you to market a year quicker. DD On 7/4/07, Simon P. Ditner [EMAIL PROTECTED] wrote: http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | The Toronto Asterisk Users Group | Join the discussion group by visiting http://taug.ca
RE: [on-asterisk] surprising google purchase
That's it! Thanks for the update. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Simon P. Ditner Sent: July 4, 2007 9:31 AM To: Mike Dancy Cc: Dave Donovan; asterisk@uc.org Subject: Re: [on-asterisk] surprising google purchase Are you thinking of Iotum's presence management suite? I think they were tightly integrating things with Exchange, but have now re-focused on the Blackberry market. On 7/4/07, Mike Dancy [EMAIL PROTECTED] wrote: Wasn't there someone in Ottawa who had a similar business? They would assign one phone number to you and it would be forwarded to various phones. I seem to recall thinking that it was quite cool so maybe Google is hoping for the same. (note that this service only applies to USA ) Though you would still need a provider whether it's a cell service, land line etc. to forward this number and voice mail to. Maybe they'll tie it in with their gmail for a total communication tool. ?? From: Dave Donovan [mailto:[EMAIL PROTECTED] Sent: July 4, 2007 9:12 AM To: asterisk@uc.org Subject: Re: [on-asterisk] surprising google purchase Sarcasm What innovation! /Sarcasm Not only is it a commodity feature on many systems like Asterisk but it's already been commercialized for years. (IE Bell's Prime Line). 50M isn't too much money, maybe they've got infrastructure in place and a staff to manage it. Just having that out of the way would get you to market a year quicker. DD On 7/4/07, Simon P. Ditner [EMAIL PROTECTED] wrote: http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- | It ain't what you don't know that gets you into trouble. It's what | you know for sure that just ain't so. -- Mark Twain | | The Toronto Asterisk Users Group | Join the discussion group by visiting http://taug.ca
Re: [on-asterisk] surprising google purchase
I agree, the admin is worth $50M (and what's $50million to Google.peanuts) Henry Dave Donovan wrote: Sarcasm What innovation! /Sarcasm Not only is it a commodity feature on many systems like Asterisk but it's already been commercialized for years. (IE Bell's Prime Line). 50M isn't too much money, maybe they've got infrastructure in place and a staff to manage it. Just having that out of the way would get you to market a year quicker. DD On 7/4/07, *Simon P. Ditner* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: http://googleblog.blogspot.com/2007/07/all-aboard.html I still can't believe it, Google bought GrandCentral for $50m! For what? Certainly not IP... - To unsubscribe, e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Re: [on-asterisk] Calls not connecting on PRI span
zaptel.conf has one pri_cpe and one pri_net
I see the calls being made but no connecting message. show channels
shows the one call in Ring state and never getting connected.
topbx*CLI show channels
Channel Location State Application(Data)
0 active channels
0 active calls
-- Executing Dial(SIP/7046-052dafd0, Zap/G0/300|45) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/300
-- Accepting call from '7046' to '300' on channel 0/23, span 2
-- Executing Answer(Zap/47-1, ) in new stack
-- Executing MusicOnHold(Zap/47-1, ) in new stack
-- Started music on hold, class 'default', on Zap/47-1
-- Zap/23-1 is proceeding passing it to SIP/7046-052dafd0
topbx*CLI show channels
Channel Location State Application(Data)
Zap/47-1 [EMAIL PROTECTED]:2 Up MusicOnHold()
Zap/23-1 [EMAIL PROTECTED]:1Dialing AppDial((Outgoing Line))
SIP/7046-052dafd0[EMAIL PROTECTED]:1RingDial(Zap/G0/300|45)
3 active channels
2 active calls
-- Nobody picked up in 45000 ms
-- Hungup 'Zap/23-1'
-- Channel 0/23, span 2 got hangup request, cause 16
topbx*CLI show channels
Channel Location State Application(Data)
Zap/47-1 [EMAIL PROTECTED]:2 Up MusicOnHold()
SIP/7046-052dafd0[EMAIL PROTECTED]:2Ring(None)
2 active channels
2 active calls
-- Timeout on SIP/7046-052dafd0
== CDR updated on SIP/7046-052dafd0
-- Executing Hangup(SIP/7046-052dafd0, ) in new stack
== Spawn extension (esi-toronto, t, 1) exited non-zero on
'SIP/7046-052dafd0'
topbx*CLI show channels
Channel Location State Application(Data)
Zap/47-1 [EMAIL PROTECTED]:2 Up MusicOnHold()
1 active channel
1 active call
topbx*CLI
/var/log/asterisk/cdr-csv/Master.csv
,7046,t,esi-toronto,device
7046,SIP/7046-1bf3ea20,Zap/23-1,Hangup,,2007-07-04
11:13:40,,2007-07-04 11:14:35,55,0,NO ANSWER,DOCUMENTATION
McQuiggan, Mark - Broadridge (Toronto) wrote:
David:
What's you zaptel.conf say? Are both groups set to pri_net or pri_cpe
signalling? You'll need that for a crossover cable...
Thanks,
Mark.
Please note: Effective April 1, my e-mail address is
[EMAIL PROTECTED]
-Original Message-
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 03, 2007 1:57 PM
To: asterisk@uc.org
Subject: [on-asterisk] Calls not connecting on PRI span
Here is an interesting support problem.
New A102D card. Span 0/1 tied with a cross over cable (to test).
Dial an extension out one span
exten = _30X,1,Dial(Zap/G0/${EXTEN},45)
Should dial out on last channel of group 0 and should arrive on last
channel of group 1 (other span)
[from-internal]
; inbound from (group 1) Nortel defaults to pass-thru to PSTN
; pstn on trunk group 0, descending order
exten = _.,1,MusiconHold()
I should hear music. However I get no connection ... Last message is
proceeding below, no ringing yet the channel (47-1) won't hangup.
It's like it doesn't know to connect the two legs of the call!!
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/7046-1bebd770,
Zap/G0/300|45) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/300
-- Accepting call from '7046' to '300' on channel 0/23, span 2
-- Executing [EMAIL PROTECTED]:1] MusicOnHold(Zap/47-1, ) in
new stack
-- Started music on hold, class 'default', on Zap/47-1
-- Zap/23-1 is proceeding passing it to SIP/7046-1bebd770
topbx*CLI
Asterisk 1.4.5
Zaptel 1.4.3
Sangoma Wanpipe 2.3.4-10
Any ideas??
dbc.
--
David Cook
-
To unsubscribe, e-mail: [EMAIL PROTECTED]
For additional commands, e-mail: [EMAIL PROTECTED]
This message and any attachments are intended only for the use of the addressee
and
may contain information that is privileged and confidential. If the reader of the
message is not the intended recipient or an authorized representative of the
intended recipient, you are hereby notified that any dissemination of this
communication is strictly prohibited. If you have received this communication in
error, please notify us immediately by e-mail and delete the message and any
attachments from your system.
RE: [on-asterisk] Calls not connecting on PRI span
Dave:
Try a straight-through cable. You need a straight-thru to connect a pri_cpe
to a pri_net.
You may want to check that the pri_net group timing is set to 1 (primary) in
your zaptel.conf, as well
(http://www.voip-info.org/wiki/index.php?page=Zaptel.conf+span+syntax).
Thanks,
M.
Please note: Effective April 1, my e-mail address is
[EMAIL PROTECTED]
-Original Message-
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Wednesday, July 04, 2007 11:23 AM
To: McQuiggan, Mark - Broadridge (Toronto)
Cc: asterisk@uc.org
Subject: Re: [on-asterisk] Calls not connecting on PRI span
zaptel.conf has one pri_cpe and one pri_net
I see the calls being made but no connecting message. show channels
shows the one call in Ring state and never getting connected.
topbx*CLI show channels
Channel Location State Application(Data)
0 active channels
0 active calls
-- Executing Dial(SIP/7046-052dafd0, Zap/G0/300|45) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/300
-- Accepting call from '7046' to '300' on channel 0/23, span 2
-- Executing Answer(Zap/47-1, ) in new stack
-- Executing MusicOnHold(Zap/47-1, ) in new stack
-- Started music on hold, class 'default', on Zap/47-1
-- Zap/23-1 is proceeding passing it to SIP/7046-052dafd0
topbx*CLI show channels
Channel Location State Application(Data)
Zap/47-1 [EMAIL PROTECTED]:2 Up MusicOnHold()
Zap/23-1 [EMAIL PROTECTED]:1Dialing AppDial((Outgoing Line))
SIP/7046-052dafd0[EMAIL PROTECTED]:1RingDial(Zap/G0/300|45)
3 active channels
2 active calls
-- Nobody picked up in 45000 ms
-- Hungup 'Zap/23-1'
-- Channel 0/23, span 2 got hangup request, cause 16
topbx*CLI show channels
Channel Location State Application(Data)
Zap/47-1 [EMAIL PROTECTED]:2 Up MusicOnHold()
SIP/7046-052dafd0[EMAIL PROTECTED]:2Ring(None)
2 active channels
2 active calls
-- Timeout on SIP/7046-052dafd0
== CDR updated on SIP/7046-052dafd0
-- Executing Hangup(SIP/7046-052dafd0, ) in new stack
== Spawn extension (esi-toronto, t, 1) exited non-zero on
'SIP/7046-052dafd0'
topbx*CLI show channels
Channel Location State Application(Data)
Zap/47-1 [EMAIL PROTECTED]:2 Up MusicOnHold()
1 active channel
1 active call
topbx*CLI
/var/log/asterisk/cdr-csv/Master.csv
,7046,t,esi-toronto,device
7046,SIP/7046-1bf3ea20,Zap/23-1,Hangup,,2007-07-04
11:13:40,,2007-07-04 11:14:35,55,0,NO ANSWER,DOCUMENTATION
McQuiggan, Mark - Broadridge (Toronto) wrote:
David:
What's you zaptel.conf say? Are both groups set to pri_net or pri_cpe
signalling? You'll need that for a crossover cable...
Thanks,
Mark.
Please note: Effective April 1, my e-mail address is
[EMAIL PROTECTED]
-Original Message-
From: David Cook [mailto:[EMAIL PROTECTED]
Sent: Tuesday, July 03, 2007 1:57 PM
To: asterisk@uc.org
Subject: [on-asterisk] Calls not connecting on PRI span
Here is an interesting support problem.
New A102D card. Span 0/1 tied with a cross over cable (to test).
Dial an extension out one span
exten = _30X,1,Dial(Zap/G0/${EXTEN},45)
Should dial out on last channel of group 0 and should arrive on last
channel of group 1 (other span)
[from-internal]
; inbound from (group 1) Nortel defaults to pass-thru to PSTN
; pstn on trunk group 0, descending order
exten = _.,1,MusiconHold()
I should hear music. However I get no connection ... Last message is
proceeding below, no ringing yet the channel (47-1) won't hangup.
It's like it doesn't know to connect the two legs of the call!!
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/7046-1bebd770,
Zap/G0/300|45) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called G0/300
-- Accepting call from '7046' to '300' on channel 0/23, span 2
-- Executing [EMAIL PROTECTED]:1] MusicOnHold(Zap/47-1, ) in
new stack
-- Started music on hold, class 'default', on Zap/47-1
-- Zap/23-1 is proceeding passing it to SIP/7046-1bebd770
topbx*CLI
Asterisk 1.4.5
Zaptel 1.4.3
Sangoma Wanpipe 2.3.4-10
Any ideas??
dbc.
--
David Cook
-
To unsubscribe, e-mail: [EMAIL PROTECTED]
For additional commands, e-mail: [EMAIL PROTECTED]
This message and any attachments are intended only for the use of the
addressee and
may contain information that is privileged and confidential. If the reader
of the
message is not the intended recipient or an authorized representative of
the
intended recipient, you are hereby notified that any dissemination of this
communication is strictly prohibited. If you have received this
communication in
error, please notify us immediately by e-mail and delete the message and
any
attachments from your system.
Re: [on-asterisk] No DTMF on conference call and COS for extensions
Sorry, no idea since I avoid Trixbox like the plague. On 6/18/07, Henry L.Coleman [EMAIL PROTECTED] wrote: Hi Leif, Silly question ...can I upgrade my Trixbox with 1.4.5 ? PS I don't need to keep the existing configuration Henry Leif Madsen wrote: Just so you guys know, I upgraded to 1.4.5 (which was released yesterday after Russell and I fixed a bug in chan_sip). I do not have my DTMF problems thus far, so I would recommend trying 1.4.5 which had a lot of DTMF things fixed post-1.4.4. Leif Madsen On 6/15/07, Stephan Monette [EMAIL PROTECTED] wrote: This is a known issue with Asterisk since it's not 100% compliant with RFC2833. We have the same problem using 1.2.x or 1.4.4. This is something to do with Variable Length DTMF tones in RFC2833. Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Leif Madsen wrote: On 6/13/07, Henry L.Coleman [EMAIL PROTECTED] wrote: 1. A three way conference call is set up by party B between A and Band C party A has to navigate party C's IVR but the DTMF signaling doesn't work. Party B is a Grandstream GXP 2000 (SIP) with signalling set to via RTP (RFC2833) The trunks are IAX (unlimitel) Hey Henry Just to let you know, I'm having the exact same problem (DTMF with RFC2833 on Asterisk, using the SIP channel). When I call into a conference and it asks for the PIN, it doesn't accept it -- the strangest thing is that Asterisk sees the DTMF via the logger.conf, console = dtmf settings. I'm still trying to track this down and will update this thread with a bug number once I get some more information. For now, I'm piggy-backing onto this bug, but I think I gotta open something separate (I don't think the issues are related anymore): http://bugs.digium.com/view.php?id=9959 More information to follow when I get it. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk
Re: [on-asterisk] No DTMF on conference call and COS for extensions
Hey Henry: Leif and me should give you a crash course on hard core Asterisk :). All the problems you are facing will go away when you get down and dirty with Asterisk CLI and codes by hand ;). Cheers! Reza. - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: TAUG - Tech asterisk@uc.org Sent: Wednesday, July 04, 2007 1:21 PM Subject: Re: [on-asterisk] No DTMF on conference call and COS for extensions Sorry, no idea since I avoid Trixbox like the plague. On 6/18/07, Henry L.Coleman [EMAIL PROTECTED] wrote: Hi Leif, Silly question ...can I upgrade my Trixbox with 1.4.5 ? PS I don't need to keep the existing configuration Henry Leif Madsen wrote: Just so you guys know, I upgraded to 1.4.5 (which was released yesterday after Russell and I fixed a bug in chan_sip). I do not have my DTMF problems thus far, so I would recommend trying 1.4.5 which had a lot of DTMF things fixed post-1.4.4. Leif Madsen On 6/15/07, Stephan Monette [EMAIL PROTECTED] wrote: This is a known issue with Asterisk since it's not 100% compliant with RFC2833. We have the same problem using 1.2.x or 1.4.4. This is something to do with Variable Length DTMF tones in RFC2833. Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Leif Madsen wrote: On 6/13/07, Henry L.Coleman [EMAIL PROTECTED] wrote: 1. A three way conference call is set up by party B between A and Band C party A has to navigate party C's IVR but the DTMF signaling doesn't work. Party B is a Grandstream GXP 2000 (SIP) with signalling set to via RTP (RFC2833) The trunks are IAX (unlimitel) Hey Henry Just to let you know, I'm having the exact same problem (DTMF with RFC2833 on Asterisk, using the SIP channel). When I call into a conference and it asks for the PIN, it doesn't accept it -- the strangest thing is that Asterisk sees the DTMF via the logger.conf, console = dtmf settings. I'm still trying to track this down and will update this thread with a bug number once I get some more information. For now, I'm piggy-backing onto this bug, but I think I gotta open something separate (I don't think the issues are related anymore): http://bugs.digium.com/view.php?id=9959 More information to follow when I get it. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Calls not connecting on PRI span
On 7/4/07, McQuiggan, Mark - Broadridge (Toronto) [EMAIL PROTECTED] wrote: Dave: Try a straight-through cable. You need a straight-thru to connect a pri_cpe to a pri_net. Maybe I misunderstand, but what you're saying doesn't seem right. I've never connected Asterisk to Asterisk but I've connected Asterisk line side to a bunch of different things (Avaya and Nortel PBXs). I've alway used a T1 crossover cable and then had one side act as NET and the other as CPE. My understanding was that the crossover cable addressed the electrical requirements and the CPE/NET settings addressed signalling requirements. Am I wrong? I would expect that if Dave had the wrong cable, he wouldn't get anything at all in terms of call setup. In fact, I would think the B and D channels wouldn't initialise. Dave Donovan
Re: [on-asterisk] Calls not connecting on PRI span
Correct you are looking for a DS1-X cable. If I recall, T1 cables are RJ48c spec, so in a crossover scenario pin1 goes to pin4 and pin 2 goes to pin5 and vice versa. If you had the wrong cable, the carrier wouldn't even come up, let alone the B or D channels. - Original Message - From: Dave Donovan To: asterisk@uc.org Sent: Wednesday, July 04, 2007 2:01 PM Subject: Re: [on-asterisk] Calls not connecting on PRI span On 7/4/07, McQuiggan, Mark - Broadridge (Toronto) [EMAIL PROTECTED] wrote: Dave: Try a straight-through cable. You need a straight-thru to connect a pri_cpe to a pri_net. Maybe I misunderstand, but what you're saying doesn't seem right. I've never connected Asterisk to Asterisk but I've connected Asterisk line side to a bunch of different things (Avaya and Nortel PBXs). I've alway used a T1 crossover cable and then had one side act as NET and the other as CPE. My understanding was that the crossover cable addressed the electrical requirements and the CPE/NET settings addressed signalling requirements. Am I wrong? I would expect that if Dave had the wrong cable, he wouldn't get anything at all in terms of call setup. In fact, I would think the B and D channels wouldn't initialise. Dave Donovan
RE: [on-asterisk] Calls not connecting on PRI span
Dave, Dave: OK. I'm a moron. I tested on my spare Sangoma and got the green lights to come up on a xover cable. I got a xover between my Avaya and my Asterisk, but I have it connected as net to net (as per voip-info.org), and it's cool. I always assumed I'm getting' old. Please ignore the prvious advice. M. Please note: Effective April 1, my e-mail address is mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] _ From: Dave Donovan [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 04, 2007 2:02 PM To: asterisk@uc.org Subject: Re: [on-asterisk] Calls not connecting on PRI span On 7/4/07, McQuiggan, Mark - Broadridge (Toronto) [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Dave: Try a straight-through cable. You need a straight-thru to connect a pri_cpe to a pri_net. Maybe I misunderstand, but what you're saying doesn't seem right. I've never connected Asterisk to Asterisk but I've connected Asterisk line side to a bunch of different things (Avaya and Nortel PBXs). I've alway used a T1 crossover cable and then had one side act as NET and the other as CPE. My understanding was that the crossover cable addressed the electrical requirements and the CPE/NET settings addressed signalling requirements. Am I wrong? I would expect that if Dave had the wrong cable, he wouldn't get anything at all in terms of call setup. In fact, I would think the B and D channels wouldn't initialise. Dave Donovan This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system.
RE: [on-asterisk] surprising google purchase
Now that's an idea - Integration with Gmail. They've got Google Talk and Gmail integrated, now they can tie the voicemail box of GTalk with the Grand Central vm box, and give you a visual voicemail interface via Gmail. AR __ Alex Robar, Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham: 905-513-8000 x 223 Fax: 905-513-8040 Toronto: 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.gearytech.com http://www.gearytech.com Strategic management of technology for business. From: Mike Dancy [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 04, 2007 10:23 AM To: Dave Donovan; asterisk@uc.org Subject: RE: [on-asterisk] surprising google purchase Wasn't there someone in Ottawa who had a similar business? They would assign one phone number to you and it would be forwarded to various phones. I seem to recall thinking that it was quite cool so maybe Google is hoping for the same. (note that this service only applies to USA ) Though you would still need a provider whether it's a cell service, land line etc. to forward this number and voice mail to. Maybe they'll tie it in with their gmail for a total communication tool. ?? -- ExchangeDefender Message Security: Click below to verify authenticity http://www.exchangedefender.com/verify.asp?id=l64IkZMK006045[EMAIL PROTECTED]
Sangoma A101 and Asterisk 1.4
Hello, I have a problem with Asterisk 1.4 and a Sangoma T1 card (A101). Basically I can't make the D channel come up no matter what I tried. I installed the Zaptel driver, the Libpri and the wanpipe driver following the instructions on Sangoma's web site: http://wiki.sangoma.com/wanpipe-linux-asterisk-install The Sangoma card is connected to an Adtran Atlas 550 phone switch which is also in our office, so I can configure it anyway I want. Both the switch and the card are set to DMS100. I let the wanpipe setup create the configuration files, including the zaptel.conf and zapata.conf. One other thing I noiced, is that the zaptel.conf file created by the Sangoma setup was showing span=1,0,0,esf,b8zs while according to the ztcfg.conf explanation from the sample file, if the clock is coming from the remote end (in my case the Adtran switch), it should be: span=1,1,0,esf,b8zs So I tried changing that as well, but it didn't make a difference. The light on the Sangoma card is green, which I assume it means that it can see the clock from my switch. If I run *ztcfg -v* it reports Zaptel Version: 1.4.3 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 24 channels configured. Also *wanrouter status* reports Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1| N/A | A101/2 | 22 | 0 | 1| EXT | 0 | Wanrouter Status: Device name | Protocol | Station | Status| wanpipe1| AFT HDLC | N/A | Connected | One interesting thing I noticed: I have another server running Asterisk 1.2and a TE110P clone card. On that system I can run *zap show channels* in the asterisk CLI and I get a list of the channels. On the Asterisk 1.4 box, it tells me the command is not valid. Am I missing a module or the command was removed ? Thanks, Liviu
RE: [on-asterisk] No DTMF on conference call and COS for extensions
LOL! It's kinda like formula 1 vs. rush hour communting. Can’t call one better than the other without considering all the factors. When you are an asterisk tweaker, trixbox is painful. When you are not so into the joy/pain of hand-coding, then TrixBox is a durn nice thing to have (or so I am told ;-) Jim -Original Message- From: Reza - Asterisk Enthusiast [mailto:[EMAIL PROTECTED] Sent: July 4, 2007 1:52 PM To: TAUG - Tech Subject: Re: [on-asterisk] No DTMF on conference call and COS for extensions Hey Henry: Leif and me should give you a crash course on hard core Asterisk :). All the problems you are facing will go away when you get down and dirty with Asterisk CLI and codes by hand ;). Cheers! Reza. - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: TAUG - Tech asterisk@uc.org Sent: Wednesday, July 04, 2007 1:21 PM Subject: Re: [on-asterisk] No DTMF on conference call and COS for extensions Sorry, no idea since I avoid Trixbox like the plague. On 6/18/07, Henry L.Coleman [EMAIL PROTECTED] wrote: Hi Leif, Silly question ...can I upgrade my Trixbox with 1.4.5 ? PS I don't need to keep the existing configuration Henry Leif Madsen wrote: Just so you guys know, I upgraded to 1.4.5 (which was released yesterday after Russell and I fixed a bug in chan_sip). I do not have my DTMF problems thus far, so I would recommend trying 1.4.5 which had a lot of DTMF things fixed post-1.4.4. Leif Madsen On 6/15/07, Stephan Monette [EMAIL PROTECTED] wrote: This is a known issue with Asterisk since it's not 100% compliant with RFC2833. We have the same problem using 1.2.x or 1.4.4. This is something to do with Variable Length DTMF tones in RFC2833. Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Leif Madsen wrote: On 6/13/07, Henry L.Coleman [EMAIL PROTECTED] wrote: 1. A three way conference call is set up by party B between A and Band C party A has to navigate party C's IVR but the DTMF signaling doesn't work. Party B is a Grandstream GXP 2000 (SIP) with signalling set to via RTP (RFC2833) The trunks are IAX (unlimitel) Hey Henry Just to let you know, I'm having the exact same problem (DTMF with RFC2833 on Asterisk, using the SIP channel). When I call into a conference and it asks for the PIN, it doesn't accept it -- the strangest thing is that Asterisk sees the DTMF via the logger.conf, console = dtmf settings. I'm still trying to track this down and will update this thread with a bug number once I get some more information. For now, I'm piggy-backing onto this bug, but I think I gotta open something separate (I don't think the issues are related anymore): http://bugs.digium.com/view.php?id=9959 More information to follow when I get it. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/885 - Release Date: 03/07/2007 10:02 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/885 - Release Date: 03/07/2007 10:02 AM
Re: [on-asterisk] No DTMF on conference call and COS for extensions
Yep, then you can work on an entirely new set of problems :) On 7/4/07, Reza - Asterisk Enthusiast [EMAIL PROTECTED] wrote: Hey Henry: Leif and me should give you a crash course on hard core Asterisk :). All the problems you are facing will go away when you get down and dirty with Asterisk CLI and codes by hand ;). -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk
Re: [on-asterisk] No DTMF on conference call and COS for extensions
First there was the Telephone... then there was Asterisk... then there was TrixBox... and then there was AsteriskNow! For all those having issues with TrixBox, may I recommend AsteriskNow? :). You may not have DTMF issues, ... and you will have time to devote to other issues to solve ;). On the serious note -- I've been told a LOT of DTMF issues have been resolved in the recent release of Asterisk. Cheers! Reza. - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'TAUG - Tech' asterisk@uc.org Sent: Wednesday, July 04, 2007 3:06 PM Subject: RE: [on-asterisk] No DTMF on conference call and COS for extensions LOL! It's kinda like formula 1 vs. rush hour communting. Can’t call one better than the other without considering all the factors. When you are an asterisk tweaker, trixbox is painful. When you are not so into the joy/pain of hand-coding, then TrixBox is a durn nice thing to have (or so I am told ;-) Jim -Original Message- From: Reza - Asterisk Enthusiast [mailto:[EMAIL PROTECTED] Sent: July 4, 2007 1:52 PM To: TAUG - Tech Subject: Re: [on-asterisk] No DTMF on conference call and COS for extensions Hey Henry: Leif and me should give you a crash course on hard core Asterisk :). All the problems you are facing will go away when you get down and dirty with Asterisk CLI and codes by hand ;). Cheers! Reza. - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: TAUG - Tech asterisk@uc.org Sent: Wednesday, July 04, 2007 1:21 PM Subject: Re: [on-asterisk] No DTMF on conference call and COS for extensions Sorry, no idea since I avoid Trixbox like the plague. On 6/18/07, Henry L.Coleman [EMAIL PROTECTED] wrote: Hi Leif, Silly question ...can I upgrade my Trixbox with 1.4.5 ? PS I don't need to keep the existing configuration Henry Leif Madsen wrote: Just so you guys know, I upgraded to 1.4.5 (which was released yesterday after Russell and I fixed a bug in chan_sip). I do not have my DTMF problems thus far, so I would recommend trying 1.4.5 which had a lot of DTMF things fixed post-1.4.4. Leif Madsen On 6/15/07, Stephan Monette [EMAIL PROTECTED] wrote: This is a known issue with Asterisk since it's not 100% compliant with RFC2833. We have the same problem using 1.2.x or 1.4.4. This is something to do with Variable Length DTMF tones in RFC2833. Thanks. Stephan Monette Unlimitel Inc. Tel.: 1 (877) 464-6638, x221 Leif Madsen wrote: On 6/13/07, Henry L.Coleman [EMAIL PROTECTED] wrote: 1. A three way conference call is set up by party B between A and Band C party A has to navigate party C's IVR but the DTMF signaling doesn't work. Party B is a Grandstream GXP 2000 (SIP) with signalling set to via RTP (RFC2833) The trunks are IAX (unlimitel) Hey Henry Just to let you know, I'm having the exact same problem (DTMF with RFC2833 on Asterisk, using the SIP channel). When I call into a conference and it asks for the PIN, it doesn't accept it -- the strangest thing is that Asterisk sees the DTMF via the logger.conf, console = dtmf settings. I'm still trying to track this down and will update this thread with a bug number once I get some more information. For now, I'm piggy-backing onto this bug, but I think I gotta open something separate (I don't think the issues are related anymore): http://bugs.digium.com/view.php?id=9959 More information to follow when I get it. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/885 - Release Date: 03/07/2007 10:02 AM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.9.14/885 - Release Date: 03/07/2007 10:02 AM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
