[asterisk-users] dahdi cannot make simaltaneous calls
Hi, I am encountering problem making concurrent calls using A sangoma card, It seems that the 2nd call get a congested or buzy,I connect via sip--asterisk--dahdi attached is the PRI debug messages -- Making new call for cref 32771 DL-DATA request Protocol Discriminator: Q.931 (8) len=42 TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator) Message Type: SETUP (5)TEI=0 Transmitting N(S)=5, window is open V(A)=5 K=7 Protocol Discriminator: Q.931 (8) len=42 TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent from originator) Message Type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)User information layer 1: A-Law (35) [18 03 a1 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE] [6c 0c 21 81 30 31 36 33 36 37 37 30 36 32] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '016367' ] [70 0b 80 30 31 39 36 35 30 31 30 32 34] Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '019650' ]q931.c:5039 q931_setup: Call 32771 enters state 1 (Call Initiated). Hold state: Idle-- Making new call for cref 32772 DL-DATA request Protocol Discriminator: Q.931 (8) len=42 TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent from originator) Message Type: SETUP (5)TEI=0 Transmitting N(S)=6, window is open V(A)=5 K=7 Protocol Discriminator: Q.931 (8) len=42 TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent from originator) Message Type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)User information layer 1: A-Law (35) [18 03 a1 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Preferred Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 Type: CPE] [6c 0c 21 81 30 31 36 33 36 37 37 30 36 32] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '016367' ] [70 0b 80 30 31 39 33 36 37 31 30 32 34] Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '019367' ]q931.c:5039 q931_setup: Call 32772 enters state 1 (Call Initiated). Hold state: Idle Protocol Discriminator: Q.931 (8) len=10 TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator) Message Type: SETUP ACKNOWLEDGE (13) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call-pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 24 (cs0, Channel Identification)q931.c:7390 post_handle_q931_message: Call 32771 enters state 2 (Overlap Sending). Hold state: Idle Protocol Discriminator: Q.931 (8) len=10 TEI=0 Call Ref: len= 2 (reference 3/0x3) (Sent to originator) Message Type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: CPE]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call-pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 24 (cs0, Channel Identification)q931.c:7104 post_handle_q931_message: Call 32771 enters state 3 (Outgoing Call Proceeding). Hold state: Idle Protocol Discriminator: Q.931 (8) len=9 TEI=0 Call Ref: len= 2 (reference 4/0x4) (Sent to originator) Message Type: RELEASE COMPLETE (90) [08 02 82 a2] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Circuit/channel congestion (34), class = Network Congestion (resource unavailable) (2) ]Received message for call 0x8cc6f80 on 0x8ca59a0 TEI/SAPI 0/0, call-pri is
[asterisk-users] Max amount of concurrent calls on and iax trunk
hi, wanted to ask if anybody has experienced setting up two asterisk 1.2 boxes connected via iax trunk. have u guys ever stress tested the trunks i.e how many concurrent calls can a trunk handle and whether codec has any effect on it. ___ -- 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] problem with e1 connection
I have a lot of these messages popping up in my mesages, E1 connection shows provisioned up active but I cant seem to be able to make a call. It was previously working before but stopped working after I did a reboot to the box this weekend. Anything I am missing out May 26 08:21:38 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:21:38 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:16 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:16 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:54 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:22:54 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:23:32 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 May 26 08:23:32 asterisk[205]: NOTICE[205]: chan_zap.c:8365 in pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1 usr/local/etc/asterisk/zapata.conf [channels] signalling=pri_cpe context=tme1_incoming group=1 callgroup=1 pickupgroup=1 priindication=outofband switchtype=euroisdn context=tme1_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=no faxdetect=no overlapdial=yes prilocaldialplan=national pridialplan=unknown channel = 1-15 channel = 17-31 signalling=pri_cpe context=tme1_incoming group=2 callgroup=2 pickupgroup=2 priindication=outofband switchtype=euroisdn context=tme1_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=yes faxdetect=incoming overlapdial=yes prilocaldialplan=national pridialplan=unknown channel = 32-46 channel = 48-62 signalling=pri_net context=md110_incoming group=3 callgroup=3 pickupgroup=3 priindication=outofband switchtype=euroisdn context=md110_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=no faxdetect=no overlapdial=yes prilocaldialplan=unknown pridialplan=unknown channel = 63-77 channel = 79-93 signalling=pri_net context=md110_incoming group=4 callgroup=4 pickupgroup=4 priindication=outofband switchtype=euroisdn context=md110_incoming amaflags=default busycount=4 callwaiting=no transfer=yes useincomingcalleridonzaptransfer=yes threewaycalling=yes callreturn=yes relaxdtmf=yes busydetect=no usecallerid=yes hidecallerid=no usecallingpres=yes echocancel=yes echocancelwhenbridged=no echotraining=yes immediate=no faxdetect=no overlapdial=yes prilocaldialplan=unknown pridialplan=unknown channel = 94-108 channel = 110-124 /usr/local/etc/zaptel.conf loadzone=my defaultzone=my span=1,1,0,ccs,hdb3,crc4 span=2,2,0,ccs,hdb3,crc4 span=3,0,0,ccs,hdb3,crc4 span=4,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 bchan=63-77 dchan=78 bchan=79-93 bchan=94-108 dchan=109 bchan=110-124 specs 1.80Ghz Dual Core with Sangoma 104, running asterisk 1.2.26.2 ontop of FreeBSD ___ -- 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] Anyone tested the new Sony Ericsson P1 phones..
Hi, Has anyone on this list tested out the new SE P1 phones ( http://www.uncrate.com/men/gear/cell-phones/sony-ericsson-p1/). It says it supports VOIP, wonder if it is working with asterisk. ___ --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] sangoma 102 and CAB-E1-RJ45BNC
Hi, sorry for the newbie hardware questions but here it goes scenario - our telco is feeding us e1 thru coax connection (unbalanced) - so the coax feed rx-tx goes to our old pabx using ericsson bp250 - what we wanted to do is to install asterisk in between hence telco--asterisk--bp250 using asterisk to power up the voip portion the problem is the we are getting crackling sound when we make calls from the old pabx extension, it seems that there is a lot of line noise due to emc. so here goes the newbie question: current setup is that from the coax we are using a balun using the given cables from sangoma will the cisco *CAB-E1-RJ45BNC *connector work on the 102 ie no need to use the balun ___ --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 card dying after 1hour
what kinda of cable are you using to connect to the e1. isit shielded or just the generic one they give with the card? On 1/28/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: Do you see anything in /var/log/messages? I am having a similar problem but I'm also getting some pci fatal error! messages. I had sangoma connect to the box and he couldn't find any config errors so we're leaning towards a hardware problem. - Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Friday, January 26, 2007 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --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 ___ --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
Re: [asterisk-users] Sangoma card dying after 1hour
try it out with shielded... we got the same problem previously, we resorted to making a short cable and wrapping it up with cooking foil and the problem appears no more On 1/29/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: We made our own, but it isn't shielded. Is there something specific to sangoma regarding cabling? We've made our own for Digium and Nortel equipment and all is well. - Jeremy -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Rosli Sukri *Sent:* Sunday, January 28, 2007 1:04 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Sangoma card dying after 1hour what kinda of cable are you using to connect to the e1. isit shielded or just the generic one they give with the card? On 1/28/07, Porier, Jeremy M. [EMAIL PROTECTED] wrote: Do you see anything in /var/log/messages? I am having a similar problem but I'm also getting some pci fatal error! messages. I had sangoma connect to the box and he couldn't find any config errors so we're leaning towards a hardware problem. - Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jon Schøpzinsky Sent: Friday, January 26, 2007 7:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Sangoma card dying after 1hour Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with zaptel 1.2.12 :) Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 12:03 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour Which asterisk versions etc etc? On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: I am running the newest version, from the sangoma wiki. Jon -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Steve Davies Sent: 26. januar 2007 10:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Sangoma card dying after 1hour On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List I am having a rather big problem with a sangoma A104 card, I just installed to replace a Digium TE410 card, that was acting up. But now we have a problem with the sangoma card. It runs great after being started, and calls proceed as normal, but after about 1 hour, it stops being able to make and receive calls. If I run wanpipemon debug, can see that the card still receives packets from the ISDN, but when I make a call, I cant see it in wanpipemon, and asterisk just responds with a: NOTICE[17240] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) I am pretty shure that this is a configuration issue, but are there anything I need to be aware of when moving from a Digium card to a sangoma card? Which wanpipe version? Anything lower than 2.3.4-4 should be upgraded as there are some resource leak fixes in that version. Regards, Steve ___ --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 ___ --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 ___ --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] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?
On 1/11/07, Michael Hamann [EMAIL PROTECTED] wrote: Hey Rosli, we are already using this feature which works quite well... Except for a bug(?) with the bristuff patches (pickupchan) that always picks up the latest ringing extension and not the extension I control via hint. It seems that it does not pick up the given sip extension (e.g. SIP/333) but the latest ringing SIP extension in general. ooo... our setup is a simple one just a single e1 line Right now we have the problem that when two phone are ringing and somebody pushes the pickup button, not the monitored call is picked up but the other one which is ringing at the same time on someone else´s extension. But I will try some patches the next days... The problem here is that the managers phone still rings on incoming calls. With the old traditional pbx, the manager was able to mute his phone and send all calls to his assistant. ok - lemme stew on this problem, might be a sneaky way to do this... But maybe I can do that with the action urls on the snom phones... hmm... I will try that and report if it works... Thanks anyway for your (and all the other) answer ... ___ --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] How to test VOIP quality?
you can test out hammer suite of products - it is quite pricey http://empirix.com On 1/10/07, Doug [EMAIL PROTECTED] wrote: I did a search: http://www.google.com/search?q=voip+quality+%28test+OR+testing%29+asterisk-users+site%3Amail-archive.com and found this: http://www.testyourvoip.com/ This seems to have quite a bit of detail. Does anyone have a better solution for testing VOIP quality? Comments? ___ --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] how to realize chief - secretary (or Manager - Assistant) setup with Asterisk?
check this out,
http://snom.com/wiki/index.php/Snom300/Web_Interface/Function_Keys#Dialog_state_.26_call_pickup
we are using this snom feature on our box, while enabling hints for the
extensions.. So basically after rebooting the snom the extension monitored
(i,e managers ext) if he is on the phone the LED will light up. If his
extension is ringing or someone is calling his extension the LED will blink.
On 1/11/07, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 10.01.2007, 16:37 +0100 schrieb Michael Hamann:
Hello,
we are running a Asterisk (1.2) installation with about 80 snom phones
(300,320,360).
Now have the demand for a special manager - assistant setup for a few
extensions.
Since Shared Line Appearance is not available in 1.2 I´m wondering how
to realize this...
What we need is that the manager can decide whether he wants to get
calls or not. If not he must have the possibility to redirect all
incoming calls to his secretary. The secretary itself answers all calls
and decides if the call is important enough to disturb the manager. If
so she/he transfers the call to the manager. So the secretary can filter
the calls for the manager...
The only way I can imagine so far is via a redirect by AstDB on the
manager extension. The managers phone has two different lines - the
official and a secret one only the secretary uses...
Or are there any other solutions?
Any hint will be appreciated ...
Hello Michael,
as I see it, the most obvious setup would be
- have SIP accounts, e.g. sip123 for the secretary phone, sip456 and
sip789 for the manager phone.
- the official/public extension number for the manager might be
4321, so
exten = 4321,1,Dial(SIP/sip123SIP/sip456)
would ring both the secretary phone and the manager phone on the
public id (which most probably can have a separate ringtone than the
private id). You would also want a private extension like
exten = 4901,1,Dial(SIP/sip789)
for the secretary to reach the manager. A few thoughts:
- The Callerid setting for both secretary and chief should be 4321, no
matter which line the chief chooses to call out through.
- Do not choose an obvious private number, like 4321 and 4322
- You could even choose a real long number, that only is available
from internal phones, and put it to a speed dial button on the secretary
phone
If you want the manager to be able to selectively not be disturbed by
public number calls, but only by his secretary, some AstDB logic could
come into the game. This can be highly dynamic, or you just configure a
few extensions by hand to do exactly this:
exten = 770/4321,1,Set(DB(list/4321)=SIP/sip123SIP/sip456)
exten = 770/4321,2,Playback(feature-donotdisturb-off)
exten = 771/4321,1,Set(DB(list/4321)=SIP/sip123)
exten = 771/4321,2,Playback(feature-donotdisturb-on)
exten = 4321,1,Dial(${DB(list/4321)})
So either the chief or the secretary could activate do-not-disturb by
dialing 771, and deactivate with 770. Just examples; choose those codes
from a range that is not in use as extensions; for my personal setup,
the 2*/3*/4*/5*/6* internal numbering for SIP devices, OOH devices, IAX
devices etc.pp., 8* being applications (like 888 the talking clock), 9*
experimental and 0* PSTN calls (how 80's! :-). A somehow similar
function (divert to VoiceMail delay in seconds can be set from any
phone, between 0 and 60 seconds) is available here as 811x.
Choose whatever suits you best.
Of course one could imagine also that the manager phone number NOT rings
the secretary while the manager is there and ready to take calls - just
edit the 770/771 lines (or add 772 for that function) - in that case,
the secretary could make use of an extension number for him/herself, but
her phone also has several lines, so why not.
HTHBR
Anselm
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Re: [asterisk-users] Secure a Asterisk Server ?
using rsync and freevrrpd would also give the same effect.. with rsync you can only mirror specific files and directory and with freevrrpd you will be creating a virtual IP and a master/slave box that does failover and failback ;) On 1/7/07, Noah Miller [EMAIL PROTECTED] wrote: actually, i have only one Asterisk Server ;=) Anyone know a how to for create a seconde asterisk in Backup for hight availability ? You can use a combination of heartbeat and drbd. Heartbeat is a clustering program that monitors your computer and its services to see if they are active. If something fails, it will automatically switch you to a backup machine. DRBD is a program that identically copies the contents of a disk from one machine to another via a dedicated network link. - Noah ___ --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] Headaches with Video over SIP
any logs/errors when you do a verbose 6 and a sip debug ?On 11/13/06, Peter Howard [EMAIL PROTECTED] wrote:On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote: On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units.And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice.But no video.And no obvious messages as to whats going wrong.The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? Do you have videosupport=yes in your sip.conf?Yes I do.I've also confirmed that I have a version of asterisk whichincludes the patch for H263P (which is what the Polycoms want to talk). --Peter HowardURSYS13 Burwood Rd,Burwood, NSW 2134Ph: 02 8745 2816Fax: 02 8745 2828___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] sip forward behind a nat
u need another box say box a with real/addressable ip address. create an iax entry in box a and have the private ip (box b) box register to box a. then you can do a Dial(IAX2/boxb/${EXTEN}) that will ring the extension connected to
your 192.168.100.249 boxhope that helps;) On 11/12/06, nik600
[EMAIL PROTECTED] wrote:Hii have to forward a call from my asterisk server on another server but
my server is behind nat.How can i setup my extension.conf?Actually i have set up it as follows:exten = 046566,1,Dial(SIP/[EMAIL PROTECTED])my server has a private ip
192.168.100.249 and doesn't have a public ipIf i try to call SIP/[EMAIL PROTECTED] from an adsl connection (with amodem, without nat) the call is routed succesfuly.If i try to forward the call from my server i cant route the call...
(i send many INVITE but without any answer)How can i fix it?many thanks in advance___--Bandwidth and Colocation provided by
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Re: [asterisk-users] Snom or Cisco Phones?
http://www.aztech.com/prod_iptelephony_ip150.htmlaztech rawks... the lcd has backlighting and methinks is snom inside ___ --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] Asterisk conferencing features
something like this in da dialplanexten = 0078,1,Answer()exten = 0078,2,Wait(2)exten = 0078,3,MeetMe(0078,idpMs)exten = 0078,4,Hangup()notes:- change 0078 to your incoming no, - so when you want to do the conference, just dial the defined extension number- or you can do a blind tranfer to the room (i.e invite)also conferencing feature is also doable on the phone, check out phones from SNOM and xlite On 10/24/06, Rafael Marangoni [EMAIL PROTECTED] wrote: Does anyone knows a simple how-to, to make sip conferencing on asterisk?2006/10/23, Rosli Sukri [EMAIL PROTECTED]: On 10/24/06, Rafael Marangoni [EMAIL PROTECTED] wrote: Hello! I'm new in Asterisk and I hope that my trouble is very simple. We're implementing a Education Project of a e-Learning system (LMS) that uses conferencing (video and audio) over internet. The e-Learning system will be on GPL license, and for that, we're using only free software to implement. Asterisk is our first choice for video and audio conferencing, and making tests, started to implement it. The questions are: 1. Asterisk makes sip conferencing? (I know the aswer is yes) yes, via the 'meet-me' application 2. Asterisk need Digium hardware to do that ? On asterisk handbook I found: Note that for technical reasons, you must have at least one Zaptel interface (of any kind) installed in your Asterisk system if you wish to use conferencing. (page 7) it needs it for 'timing'. on freebsd i have manage to install it without a physical zaptel card, by just loading the module to provide the timing 3. Asterisk make video conferencing? not yet.. it only supports video call i.e 2 party where as conference usually means more than 2 4. If yes, anyone have docs more detailed on how to do that? 5. Anyone know clients (softphones) under gpl that we can use the code to implement on this aplication? ekiga provides both audio and video capabilities, it is part of gnome. for windows you can use xlite its gratis software but not gpl I need asterisk only for internet conferencing, and I know that it's much more than that. Thanks, and sorry for the questions Rafael Marangoni ___ --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 listTo 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] Asterisk conferencing features
On 10/24/06, Rafael Marangoni [EMAIL PROTECTED] wrote: Hello!I'm new in Asterisk and I hope that my trouble is very simple.We're implementing a Education Project of a e-Learning system (LMS)that uses conferencing (video and audio) over internet.The e-Learning system will be on GPL license, and for that, we're using only free software to implement.Asterisk is our first choice for video and audio conferencing, andmaking tests, started to implement it.The questions are:1. Asterisk makes sip conferencing? (I know the aswer is yes) yes, via the 'meet-me' application2. Asterisk need Digium hardware to do that ? On asterisk handbook I found: Note that for technical reasons, you must have at least one Zaptelinterface (of any kind) installed in your Asterisk system if you wishto use conferencing. (page 7)it needs it for 'timing'. on freebsd i have manage to install it without a physical zaptel card, by just loading the module to provide the timing 3. Asterisk make video conferencing?not yet.. it only supports video call i.e 2 party where as conference usually means more than 24. If yes, anyone have docs more detailed on how to do that? 5. Anyone know clients (softphones) under gpl that we can use the codeto implement on this aplication?ekiga provides both audio and video capabilities, it is part of gnome. for windows you can use xlite its gratis software but not gpl I need asterisk only for internet conferencing, and I know that it'smuch more than that. Thanks, and sorry for the questionsRafael Marangoni___--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] echo cancellation on hard phones
Dear List,Has anybody done any tests on sip hardphones quantitatively measuring the MOS and TELR value. I am doing some type approval testing and one of the requirements is that the measured echo is = -25dB with a MOS value of 3.5. I dont know whether this -25dB value is fictitious or unachieveable or is there any settings that you need to do on asterisk side to attain that magic numberRegards Rosli ___ --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] H323
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeetingOn 8/26/06, atik khan [EMAIL PROTECTED] wrote:Hi,i used to work ooh323 with my asterisk. it gives better performance than otheroh323 or H323 comes with asterisk...i got H323 channel and oh323 with a lot of error.( like codecselection )but ooh323 works fine with methanksatikOn 26 Aug 2006 12:13:52 +0200, andrutto [EMAIL PROTECTED] wrote: Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto -- Najnowsze fakty!!! http://link.interia.pl/f1996 ___ --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 listTo 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] Port Forwarding SIP rtp
just disable iptables - if use redhat/fedora#service iptables stopOn 8/11/06, Siqhamo Sifo [EMAIL PROTECTED] wrote:I need help with SIP,RTP port forwarding , I can connect using SIP and make calls but there is no audio even though my kernel has sip support andI suspect that it has to do with iptables.Siqhamo SifoNewLunar Technology Solutions5th FloorSmartXchange5 Walnut Road Durbanhttp://www.newlunar.co.za___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Problems with Codecs in Asterisk
either1)pay digium for g.729 license or2)allow g.729 for sip3- sip 1 - sip2 work cause it will pass thru, - sip 2 - sip3 fails because since asterisk wants to do transcoding to 729-711 and no license if bandwidth is a concern just use GSM (if available as a codec on the phone)On 8/8/06, Chan Kwang Mien [EMAIL PROTECTED] wrote:Hi,My test-setup is as follows : sip1 -- Asterisk -- sip2^|--- sip3In sip.conf,[sip1]type=friendhost=dynamicsecret=passdisallow=allallow=g729allow=ulaw [sip2]type=friendhost=dynamicsecret=passdisallow=allallow=g729[sip3]type=friendhost=dynamicsecret=passdisallow=allallow=ulawsip1 supports g.729 and g.711u only sip2 supports g.729 onlysip3 supports g.711u onlysip1 is able to establish a call to sip2.However, I have problem establishing a call from sip1 to sip3. sip3rings but when I answered it, it hanged up. The Logs are :-- Executing Dial(SIP/2006-389a, SIP/2003) in new stack-- Called 2003Aug8 09:55:15 WARNING[6937]: channel.c:2725ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4) to SIP/2006-389a(256)-- SIP/2003-b5f8 is ringing-- SIP/2003-b5f8 answered SIP/2006-389aAug8 09:55:16 WARNING[6937]: channel.c:2725ast_channel_make_compatible: No path to translate from SIP/2006-389a(256) to SIP/2003-b5f8(4)Aug8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had todrop call because I couldn't make SIP/2006-389a compatible withSIP/2003-b5f8== Spawn extension (phones, 2003, 1) exited non-zero on 'SIP/2006-389a'I think the codecs used by sip3 and sip1 are incompatible. Does anyoneknow how I could make them compatible ?Thank you.Regards,Kwang Mien___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] Problems with Codecs in Asterisk
On 8/8/06, Dean @ INKnBITs [EMAIL PROTECTED] wrote: I have the same problem here, why does asterisk not use ulaw with Sip1 - Sip3 ? As it has allow=g729 and allow=ulaw in Sip1, should it not fallback onto ulaw when the g729 fails?true, it might be a problem on da sip phones itself (order of codec preference/precedence maybe) - can you confirm what codec is sip1 passing it to asterisk?.. currently for me i am using a pa1688 based sip phone and when setting the codec you have to set the precedence order. i.e try ulaw, gsm then as a last option use 729.i am speculating in this particular scenario during the initialisation of sip1 - asterisk wants bof of them probably agreed to do 729 as a result of the precedence setting on the phone maybe as an experiment, get sip3 to call sip2? Thanks, Dean. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Rosli SukriSent: 08 August 2006 13:38To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Problems with Codecs in Asteriskeither1)pay digium for g.729 license or2)allow g.729 for sip3- sip 1 - sip2 work cause it will pass thru, - sip 2 - sip3 fails because since asterisk wants to do transcoding to 729-711 and no license if bandwidth is a concern just use GSM (if available as a codec on the phone) On 8/8/06, Chan Kwang Mien [EMAIL PROTECTED] wrote: Hi,My test-setup is as follows :sip1 -- Asterisk -- sip2^|--- sip3In sip.conf,[sip1]type=friendhost=dynamicsecret=passdisallow=allallow=g729allow=ulaw[sip2]type=friendhost=dynamicsecret=passdisallow=allallow=g729[sip3] type=friendhost=dynamicsecret=passdisallow=allallow=ulawsip1 supports g.729 and g.711u only sip2 supports g.729 onlysip3 supports g.711u onlysip1 is able to establish a call to sip2.However, I have problem establishing a call from sip1 to sip3. sip3rings but when I answered it, it hanged up. The Logs are :-- Executing Dial(SIP/2006-389a, SIP/2003) in new stack-- Called 2003Aug8 09:55:15 WARNING[6937]: channel.c:2725ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4) to SIP/2006-389a(256)-- SIP/2003-b5f8 is ringing-- SIP/2003-b5f8 answered SIP/2006-389aAug8 09:55:16 WARNING[6937]: channel.c:2725ast_channel_make_compatible: No path to translate from SIP/2006-389a(256) to SIP/2003-b5f8(4)Aug8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had todrop call because I couldn't make SIP/2006-389a compatible withSIP/2003-b5f8== Spawn extension (phones, 2003, 1) exited non-zero on 'SIP/2006-389a'I think the codecs used by sip3 and sip1 are incompatible. Does anyoneknow how I could make them compatible ?Thank you.Regards,Kwang Mien___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo 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] Problems with Codecs in Asterisk
thanks radamson for the proper explanation, actually this question was also posted on the ast-dev list. I believe the issue here is that:is asterisk smart enuff to choose the proper codec over 2 sip channels and not defaulting the the ordering or preference list know how I could make them compatible ?I believe the issue is this... When sip1 initiates a call, a codec is selected based on the sip phonepreference and asterisk codec ordering. That selection has nothing todo with where the call is going to be directed (eg, sip2 or sip3). That negotiation happens early, otherwise you would not be able to hearbusy congested tones, audio messages, etc.After that negotiation happens, then asterisk begins processing thecall by doing the same thing with the destination sip phone. In other words, asterisk negotiates an appropriate codec with sip2 (or sip3) thatis based on that phone's codec preference and what asterisk's codecordering for that sip phone definition.After both of the above steps are completed, asterisk then tries to bridge the two calls, and if you don't have the g729 codec installed, itcan't bridge ulaw to g729. There is no more codec negotiation going onafter step 1 and 2 above.The above can easily be verified by simply doing a sip debug and placing a call.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] DTMF problems
test it out with rfc2833 with sip since it is the most common of them allOn 8/8/06, Moises Silva [EMAIL PROTECTED] wrote:Ok, with SIP you can send the DTMF in 3 flavors. You need to know how your SIP telephony gateway providers send and expect the DTMF. Youconfigure that in Asterisk file sip.conf, look for the peer parameterdtmfmode, valid values are:dtmfmode=infoUse SIP INFO messages to send, this is out of band dtmfmode=rfc2833Actually i dont know, but check RFC2833 :)dtmfmode=inbandThe DTMF digits are sent in the same stream that the audio. This means thatif the audio codec is of low quality, DTMF may not pass. dtmfmode=autoAsterisk is supposed to detect the correct DTMF mode to use, actuallyI havent used this one, but you can give it a try :)Regards ___ --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] Problems with Codecs in Asterisk
On 8/8/06, Chan Kwang Mien [EMAIL PROTECTED] wrote: From the SIP messages exchange, sip1 informs Asterisk in the INVITE message that it supports g.729 and g.711u. Asterisk then compares its first allowed codec which is g.729 with the supported codec by sip1. Since sip1 supports g.729 and it is an allowed codec, Asterisk chooses g.729 as the codec between itself and sip1. Asterisk then forwards the INVITE message but the codec in the INVITE is changed to g.711u. sip3 replied that it supports g.711u in the OK message. Asterisk then realised that the codec between itself and sip3 is different from the codec between itself and sip1. There is a need for transcoding. And since there isn't any g.729 Licence, the connection breaks. In short, once Asterisk is sure that the first codec of the allowed list is supported by sip1, it will use that codec and will ignore the remaining codec, in this case, g.711u. Intuitively, I thought that since sip1 supports both g.729 and g.711u, it should be able to connect to a g.729 phone or a g.711u phone via Asterisk using the same sip.conf.it can - the only problem is that it needs to do transcoding and since g.729 is proprietary and the owner wants some royalty payments from it then you are stuck in the mud I have the same problem here, why does asterisk not use ulaw with Sip1 - Sip3 ? As it has allow=g729 and allow=ulaw in Sip1, should it not fallback onto ulaw when the g729 fails? Thanks, Dean. -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ]On Behalf Of Rosli Sukri Sent: 08 August 2006 13:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems with Codecs in Asterisk either 1)pay digium for g.729 license or 2)allow g.729 for sip3 - sip 1 - sip2 work cause it will pass thru, - sip 2 - sip3 fails because since asterisk wants to do transcoding to 729-711 and no license if bandwidth is a concern just use GSM (if available as a codec on the phone) On 8/8/06, Chan Kwang Mien [EMAIL PROTECTED] wrote: Hi, My test-setup is as follows : sip1 -- Asterisk -- sip2 ^ |--- sip3 In sip.conf, [sip1] type=friend host=dynamic secret=pass disallow=all allow=g729 allow=ulaw [sip2] type=friend host=dynamic secret=pass disallow=all allow=g729 [sip3] type=friend host=dynamic secret=pass disallow=all allow=ulaw sip1 supports g.729 and g.711u only sip2 supports g.729 only sip3 supports g.711u only sip1 is able to establish a call to sip2. However, I have problem establishing a call from sip1 to sip3. sip3 rings but when I answered it, it hanged up. The Logs are : -- Executing Dial(SIP/2006-389a, SIP/2003) in new stack -- Called 2003 Aug 8 09:55:15 WARNING[6937]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4) to SIP/2006-389a(256) -- SIP/2003-b5f8 is ringing -- SIP/2003-b5f8 answered SIP/2006-389a Aug 8 09:55:16 WARNING[6937]: channel.c:2725 ast_channel_make_compatible: No path to translate from SIP/2006-389a(256) to SIP/2003-b5f8(4) Aug 8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had to drop call because I couldn't make SIP/2006-389a compatible with SIP/2003-b5f8 == Spawn extension (phones, 2003, 1) exited non-zero on 'SIP/2006-389a' I think the codecs used by sip3 and sip1 are incompatible. Does anyone know how I could make them compatible ? Thank you. Regards, Kwang Mien ___ --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 listTo 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
