Re: [asterisk-users] Asterisk Management API
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote: 8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... Ok, I'm not 100% sure if this is even possible (it should be) 1. Make a Call (Originate works fine but I can't seam to phone the voice mail using originate, or a que for that matter.) 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. Currently I'm not finding this any job any easier than the CSTA was on the Alcatel was. Peter. -- _ -- 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] Asterisk Management API
On 9 March 2010 07:58, Peter Childs pchi...@bcs.org wrote: On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote: 8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... Ok, I'm not 100% sure if this is even possible (it should be) 1. Make a Call (Originate works fine but I can't seam to phone the voice mail using originate, or a que for that matter.) Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Currently I'm not finding this any job any easier than the CSTA was on the Alcatel was. Peter. Peter. -- _ -- 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] MWI and 1.6.1
On 8 Mar 2010, at 22:08, Dave Poirier wrote: Top posting to remain consistent... I drop litter because everyone else does. ;) W -- _ -- 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] DUNDI Sip authentication failure
Hi all, I'm new in asterisk and I got to set up a dundi config for my work.
I have 2 PBX for the test, the two PBX are in the same local network
PBX A : 192.168.199.23
PBX B : 192.168.199.21
my config files : (on PBX B , the config files on PBX A looks like it)
/etc/asterisk/dundi.conf
[general]
bind=192.168.199.21
port=4520
cachetime=5
ttl=32
autokill=yes
entityid=00:30:18:4C:33:53
[mappings]
;dundi-test =
dundi-local,0,IAX2,dundi:${secr...@toronto.example.com/${NUMBER},nounsolicited,nocomunsolicit,nopartial
priv =
dundi-priv-canonical,0,SIP,dundi:${secr...@192.168.199.21/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;priv = dundi-priv-canonical,0,SIP,192.168.199.21/${NUMBER},nopartial
priv =
dundi-priv-customers,100,SIP,dundi:${secr...@192.168.199.21/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;priv = dundi-priv-customers,100,SIP,192.168.199.21/${NUMBER},nopartial
priv =
dundi-priv-customers,400,SIP,dundi:${secr...@192.168.199.21/${NUMBER},nounsolicited,nocomunsolicit,nopartial
;priv = dundi-priv-via-pstn,400,SIP,192.168.199.21/${NUMBER},nopartial
[00:40:48:B2:78:6B]
model = symmetric
host = 192.168.199.23
inkey = 192.168.199.23
outkey = 192.168.199.21
include = priv
permit = priv
qualify = yes
order = primary
*/etc/asterisk/sip_custom.conf
language=fr
nat=never
;Subscribecontext=ext-local
[priv]
type=friend
dbsecret=dundi/secret
context=dundi-priv-local
host=192.168.199.23
qualify=yes*
/etc/asterisk/extensions_custom.conf
[ext-local-custom]
;for Direct IVR dialing if IVR is installed on the PBX B
exten = _36X,1,Macro(dundi-priv,${EXTEN})
[dundi-priv-canonical]
; local number of the PBX A for dundi advertise
exten = _37X,1,Goto(ext-local,${EXTEN},1)
[dundi-priv-customers]
; If you are an ITSP or Reseller, list your customers here.
[dundi-priv-via-pstn]
; If you are freely delivering calls to the PSTN, list them here
[dundi-priv-local]
include = dundi-priv-canonical
include = dundi-priv-customers
include = dundi-priv-via-pstn
[dundi-priv-switch]
; Just a wrapper for the switch
switch = DUNDi/priv
[dundi-priv-lookup]
include = dundi-priv-local
include = dundi-priv-switch
[macro-dundi-priv]
exten = s,1,Goto(${ARG1},1)
include = dundi-priv-lookup
[trydundi]
exten = _.,1,Macro(dundi-priv,${EXTEN})
exten = _.,2,Congestion
What works : if I use (on PBX B)
dundi lookup 3...@priv
asterisk respond :
1. 0 SIP/dundi:+wxatxxjxspp8mrpal3mr...@192.168.199.23/360
(EXISTS|NOUNSLCTD|NOCOMUNSLTD)
from 00:40:48:b2:78:6b, expires in 5 s
DUNDi lookup completed in 7 ms
but if I try to call from 360 to 370 or from 370 to 360 the call fails
So it seems that I have a SIP authentication failure.
but I don't know how to find the real problem.
Can you help me ?
Here are some logs :
On the CLI prompt :
-- Executing [...@from-internal:1] ResetCDR(SIP/360-08dfe0a0, )
in new stack
-- Executing [...@from-internal:2] NoCDR(SIP/360-08dfe0a0, ) in
new stack
-- Executing [...@from-internal:3] Wait(SIP/360-08dfe0a0, 1) in
new stack
-- Executing [...@from-internal:4] Playback(SIP/360-08dfe0a0,
silence/1cannot-complete-as-dialedcheck-number-dial-again|noanswer)
in new stack
-- SIP/360-08dfe0a0 Playing 'silence/1' (language 'fr')
-- SIP/360-08dfe0a0 Playing 'cannot-complete-as-dialed' (language
'fr')
-- SIP/360-08dfe0a0 Playing 'check-number-dial-again' (language 'fr')
-- Executing [...@from-internal:5] Wait(SIP/360-08dfe0a0, 1) in
new stack
== Spawn extension (from-internal, 370, 5) exited non-zero on
'SIP/360-08dfe0a0'
-- Executing [...@from-internal:1] Macro(SIP/360-08dfe0a0,
hangupcall) in new stack
-- Executing [...@macro-hangupcall:1] ResetCDR(SIP/360-08dfe0a0, w)
in new stack
-- Executing [...@macro-hangupcall:2] NoCDR(SIP/360-08dfe0a0, ) in
new stack
-- Executing [...@macro-hangupcall:3] GotoIf(SIP/360-08dfe0a0,
1?skiprg) in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [...@macro-hangupcall:6] GotoIf(SIP/360-08dfe0a0,
1?skipblkvm) in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [...@macro-hangupcall:9] GotoIf(SIP/360-08dfe0a0,
1?theend) in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [...@macro-hangupcall:11] Hangup(SIP/360-08dfe0a0, )
in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/360-08dfe0a0' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/360-08dfe0a0'
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: NULL (Command)
Flags: 00 STrans: 29219 DTrans: 0 [192.168.199.21:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 08363 DTrans: 29219 [192.168.199.21:4520] (Final)
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: NULL (Command)
Flags: 00 STrans: 12520 DTrans: 0 [192.168.199.21:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 09513
Re: [asterisk-users] Callcenter open source program
1) elastix 2) contacq (but there is still a stable version) 2010/3/8 Edwin Quijada listas_quij...@hotmail.com gNUDIALER *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- Date: Sun, 7 Mar 2010 06:21:34 -0800 From: wassimdarwi...@yahoo.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Callcenter open source program HI all: Iam planning to use my asterisk box as callcenter ,any one can advice me with the best callcenter open source program based on asterisk . Any help will be apreciated. -- -- _ -- 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] app_queue problem with Ringing state
Hi, This is the output from queue show 28: 47 (DAHDI/g0/12345678) (realtime) (Ringing) has taken no calls yet Why is the devicestate Ringing when no channels is calling this number, and the queue says has taken no calls yet? Is it picking up the general state of a random channel on g0 in dahdi? Or what is happening? It only seems to happen with this particular queue/queuemember. And it may seem like asterisk doesn't always tro to call this channel when people enter the queue, because of the erroneous Ringing state. Even when asterisk has 0 channels in use, this state can be Ringing for this member. This is on Asterisk 1.6.1.2. Might this be a (former?) bug in device-state or app_queue? Håkon -- _ -- 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] FAX configuration for DAHDI lines
On Tue, Mar 9, 2010 at 6:37 AM, Gopalakrishnaiyer Venugopal-Q16770 venui...@motorola.com wrote: HI, Do we need to make any changes to the chan_dahdi.conf to make sure that the asterisk detects fax calls?As mentioned below I will be connecting an analog fax machine to the DAHDI channel and will be dialling that public number where fax is connected. I don't think asterisk really needs to know that it is a fax unless you have echo cancelleration on your card. But even then, dahdi will automatically disable that module without you needing to change any configuration anyways. So as far as I understand what you are trying to do, it's just to dial out as usual, no magic needed. Is the public number connected via DAHDI? Håkon -- _ -- 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] FAX configuration for DAHDI lines
Hi, Yes the public number is connected via DAHDI.Also for incoming fax do we need to make any changes? Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 * -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Håkon Nessjøen Sent: Tuesday, March 09, 2010 3:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FAX configuration for DAHDI lines On Tue, Mar 9, 2010 at 6:37 AM, Gopalakrishnaiyer Venugopal-Q16770 venui...@motorola.com wrote: HI, Do we need to make any changes to the chan_dahdi.conf to make sure that the asterisk detects fax calls?As mentioned below I will be connecting an analog fax machine to the DAHDI channel and will be dialling that public number where fax is connected. I don't think asterisk really needs to know that it is a fax unless you have echo cancelleration on your card. But even then, dahdi will automatically disable that module without you needing to change any configuration anyways. So as far as I understand what you are trying to do, it's just to dial out as usual, no magic needed. Is the public number connected via DAHDI? Håkon -- _ -- 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
Re: [asterisk-users] FAX configuration for DAHDI lines
On Tue, Mar 9, 2010 at 10:38 AM, Gopalakrishnaiyer Venugopal-Q16770 venui...@motorola.com wrote: Hi, Yes the public number is connected via DAHDI.Also for incoming fax do we need to make any changes? no -- _ -- 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] asterisk peer uses 5060 to send and 5061 to receive
Hello Im configuring an asterisk peer, wich uses port 5060 to send and port 5061 to receive signaling. So, wich port should I put in my asterisk SIP trunk configuration? port = 5060 or port = 5061 ? Thanks Regards Joao Pereira -- _ -- 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] asterisk peer uses 5060 to send and 5061 to receive
Hello Im configuring an asterisk peer, wich uses port 5060 to send and port 5061 to receive signaling. So, wich port should I put in my asterisk SIP trunk configuration? port = 5060 or port = 5061 ? Thanks Regards Joao Pereira -- _ -- 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] Disable echo canceller Fonebridge
Hello! I have problems with audio in conference zap sip, I have choppy audio. I believe this problem is cause by de echo canceller from the fonebridge that I use in my system. Can someone explain me how I can disable the echo canceller form the fonebridge? I'm using dual port T1/E1 foneBRIDGE2 Thanks!! -- _ -- 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] MWI and 1.6.1
Will Payne wrote: On 8 Mar 2010, at 22:08, Dave Poirier wrote: Top posting to remain consistent... I drop litter because everyone else does. ;) W Different entirely. People who switch to bottom posting on a top-posted thread make things MUCH harder to read by being needlessly pedantic. It's like those people who decide that, even though traffic is moving along at an average of 70mph, they're going to drive 55 in the fast lane to 'teach everyone the proper speed.' They're statistically MORE likely to cause accidents (or, in LA, get shot) than those travelling along with traffic at a speed above the posted speed limit. On some positions, it is not helpful to be unwavering. N. -- _ -- 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] MWI and 1.6.1
On 9 Mar 2010, at 11:47, SIP wrote: Different entirely. People who switch to bottom posting on a top-posted thread make things MUCH harder to read by being needlessly pedantic. it just seemed like a 'I know this is wrong, but...' comment :) Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you snip the quote down to the relevant portion, you can reply where you like, regardless of what's gone on beforehand. (Surely there's no such thing as 'needlessly' pedantic - all pedantry is necessary :) W -- _ -- 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] CallerID presented in Asterisk
Gopalakrishnaiyer Venugopal-Q16770 wrote: Caller Identity restricted. The asterisk is displaying the caller id of the caller eventhough they are not supposed to be shown. core show application setcallerpres hylafax*CLI -= Info about application 'SetCallerPres' =- [Synopsis] Set CallerID Presentation [Description] SetCallerPres(presentation): Set Caller*ID presentation on a call. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable Doug -- _ -- 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] MWI and 1.6.1
Will Payne wrote: it just seemed like a 'I know this is wrong, but...' comment :) Quoting entire emails is bad, m'kay. Quoting whole threads is worse. If you snip the quote down to the relevant portion, you can reply where you like, regardless of what's gone on beforehand. (Surely there's no such thing as 'needlessly' pedantic - all pedantry is necessary :) W Unless it's errant. Then you upset Churchill. N. -- _ -- 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] CallerID presented in Asterisk
On 9 Mar 2010, at 12:21, Gopalakrishnaiyer Venugopal-Q16770 wrote: My SIP server (SONUS) is making a call to Asterisk DAHDI line with Caller Identity restricted. The asterisk is displaying the caller id of the caller eventhough they are not supposed to be shown. Kindly throw some light on this issue FreePBX by any chance? http://www.freepbx.org/trac/ticket/3797 They know, but don't care.. Steve -- _ -- 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] Turning off DNIS on T1 set to FXO_LS protocol
On 3/8/2010 12:55 PM, Kevin P. Fleming wrote: Dean Hoover wrote: Our company has an Asterisk server where one of the T1 is connected to an IVR. Asterisk is configured for FXO Loopstart, and the IVR is configured FXS. This is under control of the dialplan, though... using Dial(DAHDI/4) without adding an extension to dial after it would cause chan_dahdi to go off hook, skip sending any digits, and go into 'answered' mode. That worked perfectly. Thank you very much. -- Dean Hoover -- _ -- 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] CallerID presented in Asterisk
Hai All, My SIP server (SONUS) is making a call to Asterisk DAHDI line with Caller Identity restricted. The asterisk is displaying the caller id of the caller eventhough they are not supposed to be shown. Kindly throw some light on this issue Regards Venugopal -- _ -- 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] Aastra, Asterisk 1.4 and Voicemail
Hi Bob, Thanks for replying. I've thought of doing that, but softkeys are limited and for a phone with many call appearances (4-5) that would be using many of the softkeys. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Bob Pierce Sent: Monday, March 08, 2010 23:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Aastra, Asterisk 1.4 and Voicemail On Mon, Mar 8, 2010 at 7:08 PM, Mike l...@virtutel.ca wrote: This seems like a basic thing to set up, so I have no doubt many people have done this. Anyone care to point me in the right direction? In our config files, we have: softkey1 type: speeddial softkey1 label: Voice Mail softkey1 value: *97 This sets up one of the Softkeys on our 480i phones as a speed dial to *97 which takes them to their voicemail. -- _ -- 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] Asterisk 1.6.2.5 crash with chan_capi upon calling to PSTN
Hi,
I am having a problem with (Asterisk is crashing) with a Fritz card PCI
/ chan_capi.
Receiving Calls from PSTN works, but outbound calls make asterisk crash
(Speicherzugriffsfehler/Segmentation fault). The crash occurs upon
dialing with the other phone not even ringing.
I hereby ask if somebody reading this list can confirm or disprove my
issue. Does anbody run a recent asterisk 2.6 with chan_capi?
Config files and backtraces are attached.
Many thanks in advance.
Daniel
Versions:
- Debian 5.0.4 Kernel 2.6.26-2-686
- Asterisk 1.6.2.5 from Digium homepage
- AVM Audiovisuelles MKTG Computer System GmbH A1 ISDN [Fritz] (rev
02)
- fritz-fcpci-src-2.6.24-2.6.28
- chan-capi-trunk (rev. 769)
/etc/asterisk/extensions.conf
-
[default]
exten = 304,1,Dial(SIP/304)
exten = 305,1,Dial(SIP/305)
;To PSTN
exten = _0.,1,Dial(CAPI/ISDN1/${EXTEN})
;From PSTN
[isdn-in]
exten = 1234567,1,Dial(SIP/304)
exten = 1234568,1,Dial(SIP/305)
/etc/asterisk/capi.conf
---
[general]
nationalprefix=0 ; or for example +49
internationalprefix=00 ; or for example +
rxgain=1.0 ;linear receive gain (1.0 = no change)
txgain=1.0 ;linear transmit gain (1.0 = no change)
language=de;set default language
[ISDN1]
isdnmode=msn ;'MSN' (point-to-multipoint,
Mehrgeraeteanschluss) or 'DID' (direct inward dial)
incomingmsn=* ;allow incoming calls to this list of MSNs/DIDs,
* = any
controller=1 ;CAPI controller number of this interface/port
group=1;dialout group
softdtmf=on;enable/disable software DTMF detection,
recommended for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed
DTMF detection
faxdetect=off ;enable faxdetection and redirection to EXTEN
'fax' for incoming and/or
faxdetecttime=0;Only detect faxes during the first 'n' seconds
of the call.
context=isdn-in;context for incoming calls
echocancelold=yes ;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
devices=2 ;number of concurrent calls (B-Channels) on this
controller
Asterisk console output
---
srvpbx:/usr/src/chan-capi-trunk# asterisk -gc Asterisk 1.6.2.5,
Copyright (C) 1999 - 2009 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com Asterisk comes with
ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General
Public License version 2 and other licenses; you are welcome to
redistribute it under certain conditions. Type 'core show license' for
details.
=
[ Booting...
[ Reading Master Configuration ]
[ Initializing Custom Configuration Options ] [Mar 9 10:17:53]
NOTICE[29368]: cdr.c:1473 do_reload: CDR simple logging enabled.
[Mar 9 10:17:53] NOTICE[29368]: loader.c:1044 load_modules: 172 modules
will be loaded.
.[Mar 9 10:17:53] NOTICE[29368]: res_smdi.c:1361 load_module: No
SMDI interfaces are available to listen on, not starting SMDI listener.
[Mar 9 10:17:53] WARNING[29368]:
chan_dahdi.c:17018 process_dahdi: Ignoring any changes to 'userbase' (on
reload) at line 23.
[Mar 9 10:17:53] WARNING[29368]: chan_dahdi.c:17018 process_dahdi:
Ignoring any changes to 'vmsecret' (on reload) at line 31.
[Mar 9 10:17:53] WARNING[29368]: chan_dahdi.c:17018 process_dahdi:
Ignoring any changes to 'hassip' (on reload) at line 35.
[Mar 9 10:17:53] WARNING[29368]: chan_dahdi.c:17018 process_dahdi:
Ignoring any changes to 'hasiax' (on reload) at line 39.
[Mar 9 10:17:53] WARNING[29368]: chan_dahdi.c:17018 process_dahdi:
Ignoring any changes to 'hasmanager' (on reload) at line 47.
[Mar 9 10:17:53] NOTICE[29368]: pbx_ael.c:122
pbx_load_module: Starting AEL load process.
[Mar 9 10:17:53] NOTICE[29368]: pbx_ael.c:135 pbx_load_module: AEL load
process: parsed config file name '/etc/asterisk/extensions.ael'.
[Mar 9 10:17:53] NOTICE[29368]: pbx_ael.c:138 pbx_load_module: AEL load
process: checked config file name '/etc/asterisk/extensions.ael'.
[Mar 9 10:17:53] NOTICE[29368]: pbx_ael.c:141 pbx_load_module: AEL load
process: compiled config file name '/etc/asterisk/extensions.ael'.
[Mar 9 10:17:53] NOTICE[29368]: pbx_ael.c:146 pbx_load_module: AEL load
process: merged config file name '/etc/asterisk/extensions.ael'.
[Mar 9 10:17:53] NOTICE[29368]: pbx_ael.c:149 pbx_load_module: AEL load
process: verified config file name '/etc/asterisk/extensions.ael'.
.[Mar 9 10:17:54] WARNING[29368]: utils.c:1536
__ast_string_field_init: trying to reset empty pool [Mar 9 10:17:54]
WARNING[29368]: utils.c:1536 __ast_string_field_init: trying to reset
empty pool [Mar 9 10:17:54] WARNING[29368]: utils.c:1536
__ast_string_field_init: trying to reset empty pool .[Mar 9
10:17:54] NOTICE[29368]: chan_skinny.c:7062
[asterisk-users] Snom Provisioning
Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy -- _ -- 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] Snom Provisioning
Hi! I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Look here: http://www.voip-info.org/wiki/view/Asterisk+phone+snom#Miscellaneous Philipp -- _ -- 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] Snom Provisioning
On Wed, Mar 10, 2010 at 3:15 AM, voip crazy voipcr...@gmail.com wrote: Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Hi have a look at the snom wiki - there is a whole section on mass deployment just need a perl script (provided) and a web server to provide files. works well Thanks. Voip-Crazy -- _ -- 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
Re: [asterisk-users] Snom Provisioning
voip crazy wrote: Hello all, I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Any clue would be welcomed. Thanks. Voip-Crazy Hi There is a setting for provisioning server for snom phones, if you set that as a server and script of your own you can set the settings remptely and also change the settings remotely, here's a good place to start http://wiki.snom.com/Functions/Phone/Mass_deployment Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- 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] Uverse, Asterisk and SIP
Fred Posner wrote: On Mar 5, 2010, at 1:01 PM, sean darcy wrote: The issues are that sip doesn't work, What does doesn't work mean? In / Out? Both? Do you have a sip trace? even though this same set up worked with POTS dsl. IAX does (but gives lousy audio quality) so I don't believe all udp ports are blocked. ifconfig on my linux router box shows the public address. I can ssh into that box from the outside. This is a dynamic address, so I use Register to set the incoming ip address. As far as I can tell the Register never gets to another asterisk box I can inspect. does your sip.conf show your external ip? I will try setting the home router address in the office asterisk box to see if that works and try a call from office to home, even though it's not a long term fix. sean With att uverse I set the firewall on the att router off, my internal router/firewall to get the public ip via dhcp (it will give you a public ip and not a private one), and then set the sip.conf parameters. ---fred And without doing anything more, it now Just Works(TM). Sunspots possibly. sean -- _ -- 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] Uverse, Asterisk and SIP
On Mar 8, 2010, at 6:16 PM, sean darcy wrote: And without doing anything more, it now Just Works(TM). Sunspots possibly. sean Glad it's working... those sunspots are nasty. :) ---fred http://qxork.com -- _ -- 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] confbridge manager/cli
I've just started switching my project to use confbridge instead of meetme and app_conference (because of audio glitches that kept appearing in those applications). However, I can't find any way to interact with an existing confbridge conference. Surely there's some equivalent to meetme's 'meetme list' command? Anything else I can use through the cli or manager API? I just need to list conferences and members. Thanks! -- Jon-o Addleman - http://www.redowl.ca -- _ -- 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] Snom Provisioning
Hi! I've to deploy about 200 snom320 phones on a instalation. Do you know any knid of tool to help me with this amount of phones? I'm thinking in a provisioning tool which I use for setting up the phones. Look here: http://www.voip-info.org/wiki/view/Asterisk+phone+snom#Miscellaneous Philipp http://wiki.snom.com/Features/Mass_Deployment -- Thanks, Phil -- _ -- 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] fax spandsp
The backtrace is not useable. Try to rebuild Asterisk with the Don't Optimize Option (make menuconfig and the the build options) regards klaus Edwin Lam wrote: Philip A. Prindeville wrote: On 03/08/2010 04:31 PM, Edwin Lam wrote: hi folks. i recently upgraded asterisk to 1.6.1.17(from 1.2) and i'm having problems with fax. after receiving fax with the ReceiveFAX app. everything seems ok. the .tiff file was there, phone line seems to hang up. then asterisk will crash. any ideas? also i looked in the log file. this is what before it crashed: [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:12] WARNING[30115] app_fax.c: WARNING T.30 ECM carrier not found [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Auto fallthrough, channel 'DAHDI/8-1' status is 'UNKNOWN' [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:1] GotoIf(DAHDI/8-1, 1?200) in new stack [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Goto (detectfax,h,200) [Mar 8 12:12:30] VERBOSE[30115] pbx.c: [Mar 8 12:12:30] -- Executing [...@detectfax:200] System(DAHDI/8-1, /usr/local/bin/mailfax /var/spool/asterisk/fax/4502-1268079069.417.tif x...@.com ) in new stack [Mar 8 12:12:31] VERBOSE[30115] chan_dahdi.c: [Mar 8 12:12:31] -- Hungup 'DAHDI/8-1' asterisk: 1.6.1.17 spandsp: 0.0.6pre17 What happens when you turn off autofallthrough? exactly same thing except instead of the Auto fallthrough line the following came up: pbx.c:3928 __ast_pbx_run: Don't know what to do with 'DAHDI/5-1' and also here's the backtracce (i'm using Debian lenny) *** glibc detected *** /usr/sbin/asterisk: double free or corruption (!prev): 0x082528b8 *** === Backtrace: = /lib/i686/cmov/libc.so.6[0xb7d66624] /lib/i686/cmov/libc.so.6(cfree+0x96)[0xb7d68826] /usr/sbin/asterisk[0x80d2e89] /lib/i686/cmov/libpthread.so.0[0xb7ce156a] /lib/i686/cmov/libc.so.6(clone+0x5e)[0xb7dd86de] -- _ -- 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] Asterisk SMDI for Nortel Option 11
Does anyone know if Asterisk can function as a voicemail system for a Nortel Option 11 PBX? We will be connecting Asterisk to act as an IVR before sending calls to the Nortel and as a Voicemail system in case the user does not answer. That part is trivial, the only problem we have is that the customer wants the Voicemail light on the Nortel phones to light up when a user gets a new message. I remember that back in 1.4 Asterisk times SMDI did not play nice with Nortel. Has this changed? Is SMDI in 1.6 compatible? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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] Queue Member stuck in Ring+InUse?
Anybody work out how to fix this? Asterisk 1.4.26.3 Sip Trunk inbound - to Queuee - Outbound to two sip stations, and one sip trunk. sip trunk caller answers, queue shows ring+inuse , core show channels shows inbound/outbound after caller hanges up, no channels in use, queue still shows ring+inuse -- _ -- 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] press release: Attrafax t.30 and t.38 alternative now released as gpl2 + commercial license.
Zoa wrote: On friday we finally released Attrafax under a GPL2 license. It comes with its own set of modems and built in transparent gatewaying. The solution should be quite stable as long as the line quality is ok. (Some tools for measuring the line quality are included in the release, as well as some fax2mail scripts). There is an example implementation included for Asterisk 1.4, if someone wants to porting it to the new fax backend or more recent asterisk versions and needs some help, let us know. Attached is an untested (I did not had the time yet) port to Asterisk 1.4.29.1 (DAHDI). Maybe the modules need some adaptions too. Maybe someone wants to give it a try. regards klaus attrafax-asterisk-1.4.29.1.patch.bz2 Description: Binary data -- _ -- 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] Which spandsp to use with 1.6.2?
Receiving a fax pstn - pstn with 1.6.2.6-rc2: -- Executing [...@incoming-pstn-line:1] Answer(DAHDI/4-1, ) in new stack -- Executing [...@incoming-pstn-line:2] Wait(DAHDI/4-1, 3) in new stack -- Executing [...@incoming-pstn-line:3] Dial(DAHDI/4-1, DAHDI/g0,36) in new stack -- Called g0 -- DAHDI/1-1 is ringing -- Redirecting DAHDI/4-1 to fax extension -- Hungup 'DAHDI/1-1' == Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on 'DAHDI/4-1' -- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax Detected) in new stack -- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1, incoming-fax,s,1) in new stack -- Goto (incoming-fax,s,1) -- Executing [...@incoming-fax:1] Set(DAHDI/4-1, FAXFILE=/var/spool/asterisk/fax/20100309_1259) in new stack -- Executing [...@incoming-fax:2] ReceiveFAX(DAHDI/4-1, /var/spool/asterisk/fax/20100309_1259.tif) in new stack [Mar 9 13:02:13] WARNING[25317]: app_fax.c:223 phase_e_handler: Error transmitting fax. result=49: The call dropped prematurely. [Mar 9 13:02:13] WARNING[25317]: app_fax.c:817 transmit: Transmission error The fax completes to a standard fax machine. I'm using spandsp-0.0.5, which, AFAICT, is the last release. However I also see: http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.6pre17.tgz and http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20100228.tar.gz Should I be using the pre-release or maybe the snapshot with 1.6.2? sean -- _ -- 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] MWI and 1.6.1
Hi Dave, Sure enough my astdb does contain references to VM files as shown with strings - doing the database dump however does not show the references. I'm not sure about the internals of how Berk DB works, however I;m also seeing references to lots of other data that really shouldn't be part of my config anymore either - like I can see some employee names that are no longer a part of our company and thus have been deleted from our * config, some several years ago. I suspect that berkdb is just not overwriting some of the data for whatever reason and has some internal mechanism for knowing what to ignore. I believe I can probably test your theory tomorrow evening though, I don't think I have too much in my astdb that can't be easily re-created, I think I can probably delete my astdb entirely and regenerate it. I'll just need to take a closer look at it first though. I would however like to believe that if * is no longer supposed to be using berkdb for any VM reference data, that any calls to read the voicemail counts from the DB should have been removed. -- Matt On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier davepoir...@gmail.com wrote: So a couple of questions I have for you Matt... If you run strings on your astdb file are you seeing references to messages files in it? #strings /var/lib/asterisk/astdb | grep -i msg and if so... If you run a db_dump185 on your astdb file do the references go away? #db_dump185 -p -f /tmp/astdb.dump astdb -- _ -- 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] CallerID presented in Asterisk
Hi Steve, So is this a bug in Asterisk 1.6? Has anyone verified/reported this issue? Warm Regards Venugopal G HNM-SO WiMAX CPE VoIP IOT Team Cell : +91-99723-99437 * -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Tuesday, March 09, 2010 6:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CallerID presented in Asterisk On 9 Mar 2010, at 12:21, Gopalakrishnaiyer Venugopal-Q16770 wrote: My SIP server (SONUS) is making a call to Asterisk DAHDI line with Caller Identity restricted. The asterisk is displaying the caller id of the caller eventhough they are not supposed to be shown. Kindly throw some light on this issue FreePBX by any chance? http://www.freepbx.org/trac/ticket/3797 They know, but don't care.. Steve -- _ -- 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] CLI not working properly - Asterisk Freez
Hello, I am using Asterisk 1.4.21.2 in a Centos 4.8 with a kernel version 2.6.9-89.ELsmp. The processor type is Intel(R) Xeon(R) Quad Core CPU E5410 @ 2.50GHz. with 4 GB of RAM Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. also i can't make any call, The output of every CLI command is weird, for example making a stop now does not work. A core show channels shows many channels but frozen, etc etc Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? My solutios is either restart the server or kill the asterisk process and restart it again Thks in advance. -- _ -- 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] call features affected by native bridging between sip phones
Hi Geeks, I am a beginner in asterisk, I read about native bridging option in asterisk which allows the RTP streaming through the SIP media terminals after initiating the call . I identified the following features are getting affected by this feature in my testing. 1) Call transfer. 2) Music On Hold 3) Conferencing with meetme. I wonder if there are any other features will get affected due to native bridging. Thanks in advance. Regards Murali Vasu -- Smile is the only priceless gift you can give without a price. -- _ -- 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
