Re: [asterisk-users] Polycom not updating the directory list
anyone? From: hin lee hi...@yahoo.com To: Asterisk Users asterisk-users@lists.digium.com Sent: Fri, March 12, 2010 10:08:53 AM Subject: Polycom not updating the directory list Hi, I have a strange problem with all of our Polycom 550 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The part that doesn't work is the MAC-directory.cfg. If I make an update to this file and reboot the phones, they do not reflect the new directory list. The only way I was able to get the phone to see the new directory list was to Format the phone. Of course this is not the ideal way. Also to add, the MAC-directory.cfg files point to 0-directory.xml. This way I only have one file to maintain. Anyone knows why it's not pull the new MAC-directory.cfg file. Thank you! -- _ -- 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] CDR: Add Dialed Number Identifierfield (DNID)field into MySQL
I have read 2 solutions
(a) Changing the Dial plan and capturing DNID and inserting it into
one of the existing column in CDR table.
(b) Copy new CDR related .c .h files which have added the
functionality of recording DNID into MySQL.
For this, CDR table structure needs to be changed and a new field has
be created in CDR table.
But I am still not very sure on how to go about doing this.
Since I only have a production server, I do not have the options of
experimenting.
Can someone help with a step-by-step?
Thx
Sanjay
On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer lee.arc...@thebigword.com
wrote:
Isn't the use of DNID separate to the userfield? I'd like to have this
working also.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: 15 March 2010 08:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
(DNID) field into MySQL
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read one big thread in the Asterisk Developer List, but I could
not figure out how to implement it ?
Is there a simple step-by-step.
If this is Asterisk 1.6.*, then you can use the adaptive ODBC, which is
configured using /etc/asterisk/cdr_adaptive_odbc.conf. If you compiled
Asterisk with samples, you will find a sample file that has pretty much
everything that you need. From there, simply set the fieldname that you
wish to write to the CDR, like this:
; Using Adaptive ODBC CDR's, sets the caller ID DNID to the CDR's custom
field named DNID
Set(CDR(DNID)=${CALLERID(DNID)})
Personally, I like to set the DNID to a variable, just in case, when the
inbound call first hits Asterisk from the trunk. This probably isn't
necessary, but I am always afraid that the CALLERID(DNID) value will change
with a transfer or a channel redirect, which we use. From there I write the
variable to the CDR.
For more information on the adaptive concept, please see
http://www.asterisk.org/node/48492. There is also more detail from Tilghman
Lesher here:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210573.html
It's very elegant in it's design and it works like a champ- we use it in
production.
If you are using Asterisk 1.4.*, you can use the the CDR's userfield. This
is an optional, user defined field that can store just about whatever data
you wish depending on the data type defined in the database. You will have
to google around to find out more information on how to enable it, although
I believe that it's an option in the /etc/asterisk/cdr.conf configuration
file that you are using.
Again, if you are using Asterisk 1.6.* I would strongly recommend that you
take advantage of the Adaptive CDR system.
I am using Asterisk 1.4.*
My cdr_mysql.conf has only the following:
[global]
hostname = localhost
dbname=asteriskcdrdb
password = amp109
user = asteriskuser
userfield=1
;port=3306
;sock=/tmp/mysql.sock
---
I could not much info on the net on this subject.
Thx
Sanjay
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Do we need an update to cdr_addon_mysql for this to work?
Lee
Still no headway.
Any help is appreciated.
Thx
Sanjay
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Re: [asterisk-users] Polycom not updating the directory list
The very obvious thing to check is the permission of the mac-addr-directory.cfg. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Thursday, 18 March 2010 4:56 PM To: Asterisk Users Subject: Re: [asterisk-users] Polycom not updating the directory list anyone? From: hin lee hi...@yahoo.com To: Asterisk Users asterisk-users@lists.digium.com Sent: Fri, March 12, 2010 10:08:53 AM Subject: Polycom not updating the directory list Hi, I have a strange problem with all of our Polycom 550 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The part that doesn't work is the MAC-directory.cfg. If I make an update to this file and reboot the phones, they do not reflect the new directory list. The only way I was able to get the phone to see the new directory list was to Format the phone. Of course this is not the ideal way. Also to add, the MAC-directory.cfg files point to 0-directory.xml. This way I only have one file to maintain. Anyone knows why it's not pull the new MAC-directory.cfg file. Thank you! -- _ -- 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] How to detect a PSTN telephone is busy or not?
hi,all one problem confuse me these days. i want to sequence dial three PSTN number(a,b,c) first, if i dial number a, if a is busy , i will dial number b. if b is busy, i will dial number c. Dial(SIP/a...@ip,30) Dial(SIP/b...@ip,30) Dial(SIP/c...@ip,30) i want to know before i dial number a, how to know if a is busy now? if a is busy now. i will not dial a, instead, i will dial number b directly. to summary is : in asterisk, how to detect a pstn telephone number is busy or not before dialing it? Thanks! -- Best regards, Sucan -- _ -- 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 fax handeling
Am 18.03.2010 05:11, schrieb Olivier: 2010/3/17 Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at Am 17.03.2010 10:40, schrieb Peter den Hartog: Hello, I was wondering if the following was possible: When somebody sends a fax to my direct number 0101234567105 (my extension will be 105) is it possible that Asterisk, or an addon sees this as a fax, and e-mail the fax to me? So everybody with a private extension will be able to receive faxes in his e-mailbox on his direct number. Yes, that should work (at least with 1.6.2): 1. Enable fax detection on the inbound channel, e.g. sip.conf: ; FAX detection will cause the SIP channel to jump to the 'fax' extension (if it exists) ; when a CNG tone is detected on an incoming call. ; ; faxdetect = yes ; Default false chan_dahdi.conf: ; For fax detection, uncomment one of the following lines. The default is *OFF* ; ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no Further, put a fax extension in the context where you handle the fax, e.g. forward to Hylafax are receive directly with ReceiveFAX(). IIRc the originaly dialed number will be written in the variable FAXEXTEN. Thus, you can send the fax to the respective email address based on FAXEXTEN. regards klaus As IMHO, fax detection needs callee to answer the incoming call, how should I proceed to get a pre-recorded audio file played (You're bout to receive an incoming fax call) to callee while the incoming fax call is converted into a fax file ? My understanding of Asterisk dialplan is : - a fax call comes in from channel A, - appropriate extension is dialed through channel B, - user answers and channels A and B are bridged, - Asterisk detects the call is a fax call and then : --- 1. stops channel B --- 2. jumps into dialplan fax priority Is it possible to play a message on channel B before stopping ? Regards makes me think that as soon as a fax is detected, the receiving channel is stopped Actually I don't know, I have never used it yet. regards klaus -- _ -- 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.
Am 17.03.2010 19:31, schrieb Matt Watson: On Tue, Mar 9, 2010 at 6:31 PM, Klaus Darilion klaus.mailingli...@pernau.at mailto:klaus.mailingli...@pernau.at wrote: 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 Just as an FYI, your 1.4.29.1 patch applies successfully against 1.4.30 as well. I've got a patched 1.4.30 system compiled and ready to install later tonight during off-hours and will begin having people test tomorrow. People here are going to be quite thrilled about having T.38 transparent gatewaying again. Fine. I had not time yet to test the 1.4.29.1 patch. FYI: For 1.6.2.6 there is also a patch available [1], based on spandsp. regards klaus [1] https://issues.asterisk.org/view.php?id=13405 -- _ -- 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] How to detect a PSTN telephone is busy or not?
Hello, Please have a look to DIALSTATUS variable. here http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUSI hope it helps On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun bit...@gmail.com wrote: hi,all one problem confuse me these days. i want to sequence dial three PSTN number(a,b,c) first, if i dial number a, if a is busy , i will dial number b. if b is busy, i will dial number c. Dial(SIP/a...@ip,30) Dial(SIP/b...@ip,30) Dial(SIP/c...@ip,30) i want to know before i dial number a, how to know if a is busy now? if a is busy now. i will not dial a, instead, i will dial number b directly. to summary is : in asterisk, how to detect a pstn telephone number is busy or not before dialing it? Thanks! -- Best regards, Sucan -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
Hello all, I would like to know if any one have experience with live audio streaming like 1. Streaming from an online resource 2. Streaming from sound card AUX interface.. What i want to accomplish is that on receiving a callers call i play back a live audio stream or stream from sound card AUX interface(It depend on caller choice). Thanks -- Best Regards Shakeel Abbas -- _ -- 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] Call Filtering
Thanks. However, I discovered a guide on doing this at the following url:- http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Example 2 shows to use a macro to present a menu to the member of staff before the call is bridged. Many thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy Sent: 17 March 2010 17:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Filtering We've done this with a bit of trickery - I believe we use System to copy a blank audio file into place before calling Dial so that Asterisk thinks the caller has already recorded their name. On 3/17/2010 1:42 PM, Dan Journo wrote: Thats similar to how I want it to work, however I dont want the caller to have to give their name (even the first time they call) Is there any way of using the p option of the dial command, but totally remove the caller name recording feature? Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin Sandy Sent: 17 March 2010 15:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Filtering Sounds like you want some type of call screening. Check out the p option to the Dial command. Hi, I would like to develop a dialplan that allows the callee to reject the call like this:- 1) Call comes in and receives a greeting and get put into a queue. 2) A second call is placed to the member of staff (SIP phone or mobile phone) 3) The member of staff answers the call and is presented with a few options. 4) If the member of staff presses 1, the incoming call is connected to the member of staff. 5) If the member of staff hangs up or presses 2, the incoming call is sent to a voicemail box. The problem being, I can't see to place the second call without bridging the first call. Can anyone point me in the right direction? Many thanks Dan -- _ -- 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] Door Phone Assistance
Yes it does. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Wednesday, March 17, 2010 8:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Door Phone Assistance Does a regular phone work on that port of the channel bank? On Wed, Mar 17, 2010 at 5:00 PM, Robert Grignon rgrig...@fleetone.com wrote: I have two Viking E10 Door phones and a Rhino FXS channel bank... I have the channel set to immediate=yes and defined a custom context... When I press the button on the door phone, the inside phone rings and I can hear the person talk through the door phone... The problem is I cant hear anything through the speaker of the door phone... I know the speaker works because I do hear the initial ringing but that's it... Could this be a voltage issue? I tried two different Viking Units... Thanks for any assistance. -- _ -- 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 -- _ -- 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] Voicemail Remote Access
Hi,
I'm trying to set up remote voicemail pickup. I've created the following
dialplan, but when I press *, I am not sent to voicemailmain. The unavailable
message just continues to play as normal.
exten = 234555,1,Set(MAILBOXID=1)
exten = 234555,n,Set(MAILBOXCONTEXT=company3)
exten = 234555,n,Voicemail(${mailbox...@${mailboxcontext},u)
exten = a,1,VoicemailMain(${mailbox...@${mailboxcontext})
Any ideas?
Thanks
Dan
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Re: [asterisk-users] SIP codec negotiation / manipulation
On 3/17/2010 6:25 PM, Jeff Brower wrote: Steve- On Wed, Mar 17, 2010 at 6:02 PM, Jeff Brower jbro...@signalogic.commailto:jbro...@signalogic.com wrote: Steve- 2010/3/17 Vinícius Fontes vinic...@canall.com.brmailto:vinic...@canall.com.br - Kevin Sandy kevin.sa...@snohio.netmailto:kevin.sa...@snohio.net escreveu: We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second invite requesting G729. However, they proceed to send us a G711 encoded audio stream... They have somewhat acknowledged the problem, but their advice is for us to only accept a single codec in our 200 OK. We don't want to disable either; we have customers using G729, so we'd like to avoid transcoding when possible, but we also do some T38 faxing, which I believe requires G711 to start off. My first thought was to selectively force the codec on inbound calls - if it is for a voice number, use 729, otherwise 711. However, I can't find any way of doing this within Asterisk. (We do have an OpenSIPS server sitting between us and the provider, and I could use OpenSIPS features to do this; however, right now the OpenSIPS server is fairly dumb - it's only proxying traffic between us and the provider and knows nothing about our specific DIDs.) A couple more details in case anyone has seen a similar issue. The provider is Broadvox, and this issue only seems to manifest on calls coming to them via Skype. They claim to not have any direct link with Skype, but it seems odd that the problem would be specific to Skype callers if the call is coming to Broadvox as a standard PSTN call. Is there any way to do this? Am I totally missing something and making a stupid mistake, or making the issue more complicated than it needs to be? If your only concern about using G711 is regarding T38, go ahead and enable G729 only. T38 doesn't need G711 at all. If your customers don't mind G729 then what is said above is fine. There will be a T.38 reinvite so it won't be G729 anymore. Canreinvite does not need to be set to yes for this to work in your sip.conf either. It can be confusing but they are different types of reinvites. I don't see how this can work if Broadvox then sends G711 anyway. I understand that to be the OP's root problem. -Jeff It doesn't matter what the codec is initially, if the provider supports T.38 and you do too, a reinvite is sent changing whatever codec over to T.38. I meant for the Broadvox voice output, but maybe your suggestion works Ok and solves his problem. -Jeff Well, I at least have more things to look into. A couple notes... 1. The provider's thought is that the reason audio is encoded incorrectly is that our 200 OK with multiple codecs is confusing their equipment. They believe that if we respond with only a single codec (which can be any of their supported codecs, and can be different per call), their equipment will handle it correctly and use the single negotiated codec. 2. I had thought there was a limit in Asterisk that it would only detect fax tones and send the re-invites for T38 if the call started as a more or less uncompressed G711 call. I may be confusing that with a limitation of some of the desktop soft-phone / fax clients I've used. It seems that at the moment, the simplest solution is going to be to setup our outbound SIP proxy as a peer in Asterisk (we're currently just hitting it by using the proxy's IP in the Dial command). We can then enable only a single codec for the outbound proxy while still allowing customer phones to use either. Unless... and I doubt it... there is some command or variable I can set before calling Answer that will modify the list of codecs we send back. That would be my ideal solution, as we could then look at the DID and decide which codec we would like for this particular call based on which codec that customer is using. -- _ -- 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] spandsp with asterisk 1.4.x
Em 17-03-2010 20:51, Vinícius Fontes escreveu: - Joao Gomes Pereiragomespere...@startel.pt escreveu: Hello Im trying to receive FAXes with my Asterisk with rxfax command. To do that, Im trying to load the app_fax.so module but asterisk says: [Mar 17 20:06:04] WARNING[11907]: loader.c:359 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Mar 17 20:06:04] WARNING[11907]: loader.c:653 load_resource: Module 'app_fax.so' could not be loaded. But I do have libspandsp.so.2 # find / -name libspandsp.so.2 /usr/local/lib/libspandsp.so.2 And yes, /usr/local/lib is in my ld.so.conf: cat /etc/ld.so.conf include ld.so.conf.d/*.conf /etc/ld.so.conf.d/*.conf /usr/local/lib /usr/include /usr/local/include What could be missing? Thanks Regards Joao Pereira Sorry for kinda hijacking your topic, but where did you get the 1.4 app_fax.so backport from? I'm really interested on that. Yes, It was difficult to find... I dont have the page, but here is the wget: wget http://downloads.sourceforge.net/project/agx-ast-addons/precompiled-linux-spandsp-app-fax.tar.bz2 Regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip: gomespere...@startel.pt -- _ -- 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] Door Phone Assistance
This is a longshot, but the FXS indication tells me you're using DAHDI. Put an answer at the start of the custom context and see if that solves your problem. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Grignon Sent: Wednesday, March 17, 2010 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Door Phone Assistance I have two Viking E10 Door phones and a Rhino FXS channel bank... I have the channel set to immediate=yes and defined a custom context... When I press the button on the door phone, the inside phone rings and I can hear the person talk through the door phone... The problem is I cant hear anything through the speaker of the door phone... I know the speaker works because I do hear the initial ringing but that's it... Could this be a voltage issue? I tried two different Viking Units... Thanks for any assistance. -- _ -- 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] Call Filtering
This is just my approach, but I would run call 1 into an AGI that produced
the second call through AMI, then proceeded based on the return.
- exten = 123,1,answer
- exten = 123,2,AGI(callproc.agi)
- exten = 123,3,Gotoif($[${PROC} = VM]?voicemail)
- exten = 123,4,Queue
- exten = 123,5(voicemail),Voicemailmain
- exten = 123,6,hangup
callproc.agi calls your staff member - if they press 1, variable proc is set
to VM.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo
Sent: Wednesday, March 17, 2010 10:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Filtering
Hi,
I would like to develop a dialplan that allows the callee to reject the call
like this:-
1) Call comes in and receives a greeting and get put into a queue.
2) A second call is placed to the member of staff (SIP phone or mobile
phone)
3) The member of staff answers the call and is presented with a few options.
4) If the member of staff presses 1, the incoming call is connected to the
member of staff.
5) If the member of staff hangs up or presses 2, the incoming call is sent
to a voicemail box.
The problem being, I can't see to place the second call without bridging the
first call.
Can anyone point me in the right direction?
Many thanks
Dan
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Re: [asterisk-users] sip send image
Do a link to the image as URL on the dial command? - exten = s,1,Dial(SIP/12345,20,KkTT,http://www.yahoo.com/image.jpg) _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bhrugu mehta Sent: Wednesday, March 17, 2010 8:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip send image Thnks for ur reply, SendImage() doesn't work with asterisk sip channel. any other solution? Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- 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] spandsp with asterisk 1.4.x
Em 17-03-2010 20:28, Doug Lytle escreveu:
Joao Gomes Pereira wrote:
What could be missing?
Running ldconfig as root
Thanks, thats it!!!
Now the module is loaded.
I just hope the FAX code works:
[macro-faxreceive]
exten = s,1,Set(FAXFILE=/var/spool/asterisk/fax/${CALLEDFAX}/${UNIQUEID})
exten = s,2,NoOP()
exten = s,3,NoOP()
exten = s,4,rxfax(${FAXFILE}.tif)
exten = s,103,Set(extmail...@startel.pt)
exten = s,104,Goto(4)
exten = s,105,Set(EXTNAME=Unknown)
exten = s,106,Goto(4)
exten = s,107,Set(EXTCOMPANY=Company)
exten = s,108,Goto(4)
Thanks again
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespere...@startel.pt
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[asterisk-users] Software for my laptop to send Fax via H.323 ?
I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323. Trying to find a way I could use my laptop to send a fax over H323 to the BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface on a Cisco router and setup a h323 dial peer to the BrookTrout, but I didn't the router with me! - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- 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] spandsp with asterisk 1.4.x
- Joao Gomes Pereira gomespere...@startel.pt escreveu: Em 17-03-2010 20:51, Vinícius Fontes escreveu: - Joao Gomes Pereiragomespere...@startel.pt escreveu: Hello Im trying to receive FAXes with my Asterisk with rxfax command. To do that, Im trying to load the app_fax.so module but asterisk says: [Mar 17 20:06:04] WARNING[11907]: loader.c:359 load_dynamic_module: Error loading module 'app_fax.so': libspandsp.so.2: cannot open shared object file: No such file or directory [Mar 17 20:06:04] WARNING[11907]: loader.c:653 load_resource: Module 'app_fax.so' could not be loaded. But I do have libspandsp.so.2 # find / -name libspandsp.so.2 /usr/local/lib/libspandsp.so.2 And yes, /usr/local/lib is in my ld.so.conf: cat /etc/ld.so.conf include ld.so.conf.d/*.conf /etc/ld.so.conf.d/*.conf /usr/local/lib /usr/include /usr/local/include What could be missing? Thanks Regards Joao Pereira Sorry for kinda hijacking your topic, but where did you get the 1.4 app_fax.so backport from? I'm really interested on that. Yes, It was difficult to find... I dont have the page, but here is the wget: wget http://downloads.sourceforge.net/project/agx-ast-addons/precompiled-linux-spandsp-app-fax.tar.bz2 Thanks a lot! Too bad it requires Zaptel instead of DAHDI. :( -- _ -- 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 DIES with no trace. PLEASE
Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance Message: 10 Date: Thu, 18 Mar 2010 00:21:06 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE HELP! To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003172121s52a3b386k5210ce03e7196...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Your configs seem good. It is hard to guess why you are having this problem. You'll need to get help from Sangoma as it is their hardware and they'll be able to check if it is a driver related issue. -- _ -- 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] Voicemail Remote Access
Dan Journo wrote: Hi, Any ideas? I'd be helpful to see the console output. 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] Voicemail Remote Access
Its ok, I discovered the issue. The DTMP signals weren't being received. All sorted now. Thanks Dan -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: 18 March 2010 14:10 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voicemail Remote Access Dan Journo wrote: Hi, Any ideas? I'd be helpful to see the console output. 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 -- _ -- 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] SIP Router Project
Hello, This Friday on VUC, the SIP Router Project, Kamailio 3.0 will be discussed with a couple experts. Your questions are welcome, as always. See the site: http://vuc.me for ways to phone in. For the best sound, use g722 and call 200...@login.zipdx.com at 12 Noon Eastern. See http://vuc.me/next for exact time in your area. Reminder too that the VoIP Users Conference will be exactly three years running on Friday March 26th. We'll be doing a special 24 hour Voipathon. Those of you in the Southern Hemisphere and Asia who are usually not awake at the time of VUC are cordially invited to join in the discussion, which will be much wider than VoIP geekdom: http://voipathon.org for more info on that. We've had some great discussions on VUC and hope to continue to do so. Please consider joining us as we are a true community, not just a podcast. We meet weekly and the talk is live. IRC channel : #vuc /r -- _ -- 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 DIES with no trace. PLEASE
Do you properly hang up the calls. Does 'zap show channel channel number' shows that the channel is 'on hook' after its hang up? On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance Message: 10 Date: Thu, 18 Mar 2010 00:21:06 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE HELP! To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003172121s52a3b386k5210ce03e7196...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Your configs seem good. It is hard to guess why you are having this problem. You'll need to get help from Sangoma as it is their hardware and they'll be able to check if it is a driver related issue. -- _ -- 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] Asterisk DIES with no trace. PLEASE
Was there any hardware upgrade in December after which you recompiled libpri? On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance Message: 10 Date: Thu, 18 Mar 2010 00:21:06 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE HELP! To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003172121s52a3b386k5210ce03e7196...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Your configs seem good. It is hard to guess why you are having this problem. You'll need to get help from Sangoma as it is their hardware and they'll be able to check if it is a driver related issue. -- _ -- 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] Setting up RTP to flow between endpoints directlybypassingAsterisk
Hi Jeff! Looks like the term native bridging is a bit overloaded. Some text from channel.h: -# When the call is answered, Asterisk bridges the media streams so the caller on the first channel can speak with the callee on the second, outbound channel -# In some cases where we have the same technology on both channels and compatible codecs, a native bridge is used. In a native bridge, the channel driver handles forwarding of incoming audio to the outbound stream internally, without sending audio frames through the PBX. -# In SIP, theres an external native bridge where Asterisk redirects the endpoint, so audio flows directly between the caller's phone and the callee's phone. Signalling stays in Asterisk in order to be able to provide a proper CDR record for the call. See also http://lists.digium.com/pipermail/asterisk-dev/2010-March/043052.html klaus Am 17.03.2010 23:34, schrieb Jeff Brower: Klaus- Am 16.03.2010 01:42, schrieb Jeff Brower: Vikram- http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly The link above indicates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing through a SIP+rtpproxy server that can do transcoding ? This would be 'native bridging' mode as I've seen it described a few places on the web, correct? If Asterisk is out of the RTP loop, then what can it still do? Only billing? It would not detect DTMF, no RTP record or announcement playout, etc. No, this this is not native bridging. Asterisk supports 3 methods of media handling: 1. bridging: media (audio, video) is received on one channel, handled over to Asterisk's core, forwarded to the bridged channel, and sent out again. 2. native-bridging: if both bridged channels use the same technology then media can be bridged directly in the channel driver, no need to feed the media into Asterisk's core. For example SIP-to-SIP calls or DAHDI-to-DAHDI calls. This is not what I understood initially from the Digium / voip-info.org web pages. For example in the SIP-to-SIP case, are you saying that still the motherboard NIC would be used and the Linux kernel would touch every packet, but Asterisk software would not? My understanding was that RTP would flow direct between the NICs on the devices. 3. bypass: here, the media flow bypasses Asterisk directly. AFAIK this works only with SIP as channel technology. This comes in 2 flavors: 3a) During call setup the media will be forwarded via Asterisk. Once the call is set-up, Asterisk will send reINVITEs to both clients using the clients original SDP contact information. For this you must set canreinvite=yes (1.4) or directmedia=yes (1.6) in sip.conf. Of course Asterisk will initiate the direct media only if the media is not needed in Asterisk, e.g. if you monitor a call, the media will always be routed via Asterisk. Ok this is what I was expecting. I thought that canreinvite=yes was equivalent to native bridging, but evidently there is a distinction here that I need to study. 3b) Media will bypass Asterisk from the beginning. Therefore you have to set directrtpsetup=yes. This is still experimental and causes weird reINVITEs (e.g. after call setup to lock down on a certain codec or after call termination to redirect media to Asterisk before hanging up). Both bypass modes Note only work if either there are no NATs at all, or the clients are behind the same NAT and do not use STUN. Ok. Thanks for this info. I was not aware of directrtpsetup. -Jeff -- _ -- 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] Setting up RTP to flow between endpoints directlybypassingAsterisk
Klaus- Looks like the term native bridging is a bit overloaded. Some text from channel.h: -# When the call is answered, Asterisk bridges the media streams so the caller on the first channel can speak with the callee on the second, outbound channel -# In some cases where we have the same technology on both channels and compatible codecs, a native bridge is used. In a native bridge, the channel driver handles forwarding of incoming audio to the outbound stream internally, without sending audio frames through the PBX. -# In SIP, theres an external native bridge where Asterisk redirects the endpoint, so audio flows directly between the caller's phone and the callee's phone. Signalling stays in Asterisk in order to be able to provide a proper CDR record for the call. See also http://lists.digium.com/pipermail/asterisk-dev/2010-March/043052.html Yes seems so. Many layers of subtlety :-) -Jeff Am 17.03.2010 23:34, schrieb Jeff Brower: Klaus- Am 16.03.2010 01:42, schrieb Jeff Brower: Vikram- http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly The link above indicates that it is possible to setup RTP streams to directly flow between endpoints and completely bypass Asterisk. I would like to know if this configuration would work when, a) both endpoints are behind NAT, and/or b) both endpoints don't support same codecs with media flowing through a SIP+rtpproxy server that can do transcoding ? This would be 'native bridging' mode as I've seen it described a few places on the web, correct? If Asterisk is out of the RTP loop, then what can it still do? Only billing? It would not detect DTMF, no RTP record or announcement playout, etc. No, this this is not native bridging. Asterisk supports 3 methods of media handling: 1. bridging: media (audio, video) is received on one channel, handled over to Asterisk's core, forwarded to the bridged channel, and sent out again. 2. native-bridging: if both bridged channels use the same technology then media can be bridged directly in the channel driver, no need to feed the media into Asterisk's core. For example SIP-to-SIP calls or DAHDI-to-DAHDI calls. This is not what I understood initially from the Digium / voip-info.org web pages. For example in the SIP-to-SIP case, are you saying that still the motherboard NIC would be used and the Linux kernel would touch every packet, but Asterisk software would not? My understanding was that RTP would flow direct between the NICs on the devices. 3. bypass: here, the media flow bypasses Asterisk directly. AFAIK this works only with SIP as channel technology. This comes in 2 flavors: 3a) During call setup the media will be forwarded via Asterisk. Once the call is set-up, Asterisk will send reINVITEs to both clients using the clients original SDP contact information. For this you must set canreinvite=yes (1.4) or directmedia=yes (1.6) in sip.conf. Of course Asterisk will initiate the direct media only if the media is not needed in Asterisk, e.g. if you monitor a call, the media will always be routed via Asterisk. Ok this is what I was expecting. I thought that canreinvite=yes was equivalent to native bridging, but evidently there is a distinction here that I need to study. 3b) Media will bypass Asterisk from the beginning. Therefore you have to set directrtpsetup=yes. This is still experimental and causes weird reINVITEs (e.g. after call setup to lock down on a certain codec or after call termination to redirect media to Asterisk before hanging up). Both bypass modes Note only work if either there are no NATs at all, or the clients are behind the same NAT and do not use STUN. Ok. Thanks for this info. I was not aware of directrtpsetup. -Jeff -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with forwarding: Now forwarding SIP/ XX to Local/
Hello, here my achitecture: client1--Asterisk1ser1---centile client2-- client1 do a call to centile. centile do a forward to client2 (Diversion) and then use the same CALL-ID! when asterisk1 receive the call with the same CALL-ID, it screen Now forwarding SIP/ -02f6 to 'Local/m...@kamailio ' (thanks to SIP/YYY-02f7) I don't want that asterisk receive the call in local because I can't read headers in local... anyone have a solution to accept a call with the same CALL-ID in SIP channel ? thank you for all, regards, -- Alexandre Rendour Acropolis Telecom http://www.acropolistelecom.net Direct: +33 (0) 181813201 Support: +33 (0) 811 851 851 rend...@acropolistelecom.net mailto:rend...@acropolistelecom.net Adresse : 161-163 avenue Gallieni Paris - Porte de Bagnolet 93170 Bagnolet -- _ -- 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] Polycom DHCP Option 66 FTP provisioning
- Original Message - From: Lee, John (Sydney) john@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 17, 2010 9:50 PM Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning I'll see if E4Strategies can open a support ticket at Polycom. They're really good about stuff like that. I'll let you know either way. What is E4Strategies? Polycom support is hopeless in Oz. They just shove you to some distributer who only knows to replace your hardware. e4strategies is the where I bought all of my Polycom hardware. It would seem that e4 moves enough Polycom hardware so P pays attention to e4 when they open a support ticket, [which is why] I (and others) buy Polycom from e4, which is why P pays attention to e4 , which is why others buy from e4... Loop: goto [which is why] :-) Preliminarily it appears that option 66 will only work with TFTP, but I'll follow up with whether or not that's 'official' or de-facto. -Karl -- _ -- 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] Polycom DHCP Option 66 FTP provisioning
On Thursday 18 March 2010 11:24:18 am Karl Fife wrote:
- Original Message -
From: Lee, John (Sydney) john@compuware.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, March 17, 2010 9:50 PM
Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
I'll see if E4Strategies can open a support ticket at Polycom.
They're really good about stuff like that. I'll let you know either
way.
What is E4Strategies?
Polycom support is hopeless in Oz. They just shove you to some
distributer who only knows to replace your hardware.
e4strategies is the where I bought all of my Polycom hardware.
It would seem that e4 moves enough Polycom hardware so P pays attention to
e4 when they open a support ticket, [which is why] I (and others) buy
Polycom from e4, which is why P pays attention to e4 , which is why others
buy from e4... Loop: goto [which is why] :-)
Preliminarily it appears that option 66 will only work with TFTP, but I'll
follow up with whether or not that's 'official' or de-facto.
-Karl
I know for a fact that you can provision a Polycom via ftp. I've included
much of my dhcpd.conf file below. Pick out what you need. Let me know if
you have questions or further difficulty.
==
ddns-update-style ad-hoc;
option subnet-mask 255.255.255.0;
option netbios-name-servers 10.0.1.1;
option domain-name-servers 208.67.222.222;
option subnet-mask 255.255.255.0;
option boot-server code 66 = string;
option time-servers pool.ntp.org;
subnet 10.0.1.0 netmask 255.255.255.0 {
option broadcast-address 10.0.1.255;
option routers 10.0.1.1;
option tftp-server-name 10.0.1.1;
range 10.0.1.50 10.0.1.60;
allow unknown-clients;
authoritative;
one-lease-per-client off;
}
# Polycom phones
group {
option boot-server ftp://polycom:pas...@10.0.1.1;;
option tftp-server-name ftp://polycom:pas...@10.0.1.1;;
option time-offset -25200;
host 0004f2278ff8 {
hardware ethernet 00:04:F2:27:8F:F8;
}
host 0004f22afafd{
hardware ethernet 00:04:F2:2A:5A:FD;
}
}
--
Take care and have fun,
Mike Diehl.
--
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Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
I'm not having problems with hanging up the calls, my problems is that i asterisk dies, i'm using a pri and always zap show channel X will always show Hookstate (FXS only): Onhook beacuse it only applies to FXS and i'm using digital e1 trunk Or am i wrong? Message: 1 Date: Thu, 18 Mar 2010 11:20:38 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Do you properly hang up the calls. Does 'zap show channel channel number' shows that the channel is 'on hook' after its hang up? On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks 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
Re: [asterisk-users] Asterisk DIES with no trace. PLEASE
E1 channels are also zap channels. Zap show channels doesn't differentiate between them. On 2010-03-18 2:05 PM, Danny Dias ing.diasda...@gmail.com wrote: I'm not having problems with hanging up the calls, my problems is that i asterisk dies, i'm using a pri and always zap show channel X will always show Hookstate (FXS only): Onhook beacuse it only applies to FXS and i'm using digital e1 trunk Or am i wrong? Message: 1 Date: Thu, 18 Mar 2010 11:20:38 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re: [asterisk-users] Asterisk DIES with no ... To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-us...@lists.digium 5ad99e891003180820l56882922gd84102d158918...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Do you properly hang up the calls. Does 'zap show channel channel number' shows that the chann... On 2010-03-18 10:06 AM, Danny Dias ing.diasda...@gmail.com wrote: Thanks Zeeshan, SAng... -- _ -- 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] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian -- _ -- 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] SPA3102 5.1.7 Firmware (codec bug in 5.1.10 ?)
On 03/18/10 16:22, Sebastian Milioto wrote: Somebody has 5.1.7 firmware for SPA3102? I'm having issues with inbound/outbound calls using asterisk through SPA3102 with firmware 5.1.10. I've read it has a codec bug, since it doesn't care about what you set up in Preferred Codec. Any help will be appreciated. Sebastian You will find it here: http://prov.802.cz/fw/ Ever since the Linksys took over from Sipura and now by Cisco, thoese devices are of very poor quality. Two of SPA3102 died on me within two years, in addition lots of echo impossible to eliminate. I've switched/replaced my Linksys/Sipura units with Audiocodes MP-114 but they are not perfect either. Though, I can say they don't have/generate any echo problems and fixes go through without any problem (which I can not say the same about Linksys/Sipura units.) -- Joseph -- _ -- 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] Free Daily Asterisk News iPhone and iPod Touch app
Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Live Audio Streaming- From Aux interface-Online resource
I would like to know if any one have experience with live audio streaming like 1. Streaming from an online resource Look at app_ices and icecast. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices 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] queue MOH
On 15/03/10 11:23 AM, Thomas Perron wrote: I want callers to enter a queue and then hear music on hold. does anyone have notes on how to integrate queuing to a dial plan that uses moh? You can just set the music on hold class for the Queue in queues.conf - you actually have to provide an option (r IIRC) to provide ringing instead of music on hold. -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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 running on a Xen Centos Server challenge!!!
Hi David! Thanks very much for helping me out will all ! Ok i try your tip and @ the moment i still have the same problem but now i have the kernel and the kernel devel the same but wend i try to run make i still get the same erro, do you guys have any idea how to fix it? -bash-3.2# rpm -qa kernel* kernel-xen-devel-2.6.18-164.6.1.el5 kernel-xen-2.6.18-164.6.1.el5 -bash-3.2# Please! Thanks very much. Daniel Abreu. On 17 Mar 2010, at 12:29 PM, David Backeberg wrote: On Wed, Mar 17, 2010 at 10:16 AM, Daniel Leite de Abreu dlab...@gmail.com wrote: -bash-3.2# cd dahdi-linux-complete-2.2.1+2.2.1/ -bash-3.2# make all make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-164.6.1.el5xen kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.2.1+2.2.1/linux' make: *** [all] Error 2 This error tells me that i don't have the sources for the kernel 2.6.18-164.6.1.el5xen , so how can i find it? http://wiki.centos.org/HowTos/I_need_the_Kernel_Source -- _ -- 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] Free Daily Asterisk News iPhone and iPod Touch app
Thanks Matt. This should be useful. I'll give it a try on my Motorola Droid/Milestone. On 2010-03-18 5:31 PM, Matt Riddell li...@venturevoip.com wrote: Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Free Daily Asterisk News iPhone and iPod Touch app
On 2010-03-18 5:31 PM, Matt Riddell li...@venturevoip.com wrote: Hi all, I've released another free app for the iPhone and iPod touch - this one lets you read the Daily Asterisk News. Hope you enjoy it :D http://www.venturevoip.com/news.php?rssid=2371 -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Asterisk running on a Xen Centos Server challenge!!!
On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote: Hi David! Thanks very much for helping me out will all ! Ok i try your tip and @ the moment i still have the same problem but now i have the kernel and the kernel devel the same but wend i try to run make i still get the same erro, do you guys have any idea how to fix it? -bash-3.2# rpm -qa kernel* kernel-xen-devel-2.6.18-164.6.1.el5 kernel-xen-2.6.18-164.6.1.el5 -bash-3.2# After you install the kernel source, you'll need to rerun ./configure. You may want to run make clean and / or make distclean before rerunning ./configure. -- Thanks, --Warren Selby http://www.selbytech.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
Re: [asterisk-users] Asterisk running on a Xen Centos Server challenge!!!
On Thu, Mar 18, 2010 at 05:03:12PM -0500, Warren Selby wrote: On Thu, Mar 18, 2010 at 6:56 PM, Daniel Leite de Abreu dlab...@gmail.comwrote: Hi David! Thanks very much for helping me out will all ! Ok i try your tip and @ the moment i still have the same problem but now i have the kernel and the kernel devel the same but wend i try to run make i still get the same erro, do you guys have any idea how to fix it? -bash-3.2# rpm -qa kernel* kernel-xen-devel-2.6.18-164.6.1.el5 kernel-xen-2.6.18-164.6.1.el5 -bash-3.2# After you install the kernel source, you'll need to rerun ./configure. Nope. The dahdi-linux makefile has no ./configure . You may want to run make clean and / or make distclean before rerunning ./configure. Nope. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hello, I'm looking for some advice on securing Asterisk. Recently my servers been under several brute-force SIP attacks. I have several remote sites, as well as many roaming users, who may have PC softclients and/or SIP based hardphones. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? I haven't looked at Secure SIP in quite a while, is that now integrated into 1.6 ? One thing that's confusing me in my config, is that I thought that if I set NAT=no in sip.conf, then I wouldn't be able to connect to that SIP account unless I was on the local LAN, specified by locallan= However in some testing, I'm finding that I can still connect from an external SIP client. Also, I tried setting one SIP account from host=dynamic to host=ipaddr, and when that client tried to register, then Asterisk complained that the account wasn't supposed to be trying to register. My next step is also to upgrade my Asterisk itself up to the latest stable 1.6 Any other suggestions? Thanks, Adrian -- _ -- 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] (no subject)
On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asterisk. Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] (no subject)
On Fri, 19 Mar 2010, Adrian Marsh wrote: I’m looking for some advice on securing Asterisk. My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Asterisk doesn't differentiate between a hard phone and a soft phone. You can restrict by IP address for soft phones as well. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? I'm a 1.2 Luddite, so I can't speak for 1.6. I think any brute force or DOS security policy needs to be implemented external to Asterisk. I don't think there are any AMI events you could listen to. I think you are limited to what you can scrounge out of a log file. How about setting up a couple of honey-pot SIP accounts with obvious passwords and in the context fire off a user event? Then you could listen for the event via AMI. Any other suggestions? Repost with a meaningful subject -- a blank subject labels you as a newbie who is probably not worth the time of members with relevant experience. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000-- _ -- 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] (no subject)
Fail2ban is a must. I was a victim of such attacks, and have implemented some other measures too, but fail2ban is a must have with the link posted by Matt which describes how to set it up for asterisk. Make sure you put your own ip address in ignore list otherwise it can block you too. On 2010-03-18 8:45 PM, Matt Riddell li...@venturevoip.com wrote: On 19/03/10 1:19 PM, Adrian Marsh wrote: Hello, I’m looking for some advice on securing Asteri... Have a look at fail2ban: http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk -- Cheers, Matt Riddell Managing Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/exchange.php (Full ITSP Solution) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) -- _ -- 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] Define an array of sip number in sip.conf
Hi List, How can I define an array of sip number in sip.conf ? I want to define an array of sip number from 1000 to 2000, so I can make a performance test on Asterisk using sipp. Thanks in Advance, Giangnh -- _ -- 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] Define an array of sip number in sip.conf
You'll have to type them all in manually. Or do what I did several times, write a script in php which will generate the sip.conf with that many extensions. Even better look into using realtime architecture, where you can quickly generate as many extensions as you like. On 2010-03-18 10:09 PM, huu giang huugiang...@yahoo.com wrote: Hi List, How can I define an array of sip number in sip.conf ? I want to define an array of sip number from 1000 to 2000, so I can make a performance test on Asterisk using sipp. Thanks in Advance, Giangnh -- _ -- 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] Better SIP security please! Was: (no subject)
Hey hey! My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Asterisk doesn't differentiate between a hard phone and a soft phone. Although: One could think about enhancing Asterisk security by allowing only a (number of) specific SIP user agent header (vendor, model) for a SIP account - next to a strong password, of course. Or implement something more dynamic like: Read and lock the current (or first) user agent string, and then ping the admin if that changes and request an un- lock/re-auth. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? It would still be important to have a sip.conf paramter in 1.4 that is similar to delayreject in iax.conf! One of my system has been scanned 3 times in the past days, and it takes just a little over a minute for a 10.000 account registration scan. 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] Better SIP security please! Was: (no subject)
Philipp, remembering sip user agent is a wondeful idea, and if you goggle it, somebody had made a patch for it, so that one could identify sip devices by their sip user agent names. Surprisingly the decision makers didn't like to put it in the production branch of asterisk at that time, however it is still avialble online somewhere as a patch for older releases of asterisk. I came across it when hackers where attacking my server on constant basis. I however ended up writing a security code within the dialplan to catch the sip user agent fields and ip addresses and compare them with info in the actual user database, which worked good for me. Here the only problem could be with change of sip user agent info, e.g. x-lite puts version number in sip user agent field, which changes as you upgrade it to newer versions. A relatively more complicated code probably will however recognize it. And a hacker can always send a fake sip user agent field if he is really desparate to hack your server, which can also be caught using fail2ban. On 2010-03-18 10:45 PM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hey hey! My first step will be to strengthen the passwords in use, and for the hardphones to restrict by IP address, but that still leaves the softphone quite widely open. Asterisk doesn't differentiate between a hard phone and a soft phone. Although: One could think about enhancing Asterisk security by allowing only a (number of) specific SIP user agent header (vendor, model) for a SIP account - next to a strong password, of course. Or implement something more dynamic like: Read and lock the current (or first) user agent string, and then ping the admin if that changes and request an un- lock/re-auth. Does Asterisk 1.6 have anything in it that can automatically block out an attacking IP, say if it receives several 20 or so failed attempts from that IP in x minutes? It would still be important to have a sip.conf paramter in 1.4 that is similar to delayreject in iax.conf! One of my system has been scanned 3 times in the past days, and it takes just a little over a minute for a 10.000 account registration scan. 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 -- _ -- 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 not working?
Hi guys,
I'm trying to move away from meetme to loose the dependency on dahdi.
ConfBridge seems to be a good fit but I can't get it going. The document
sounds like an easy to use app. Am I missing any bridge_ modules?
Asterisk 1.6.2.0~rc2-0ubuntu1.2
-- Executing [...@outbound:1] Answer(SIP/109-b877a8c8, ) in new
stack
-- Executing [...@outbound:2] ConfBridge(SIP/109-b877a8c8,
conf) in new stack
[Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434
join_conference_bridge: Conference bridge '521' could not be created.
dial plan:
exten = _52X,1,Answer()
exten = _52X,n,ConfBridge(${EXTEN})
Thanks,
Kelvin
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Re: [asterisk-users] How to detect a PSTN telephone is busy or not?
Thanks! but if i use Queue to call out not Dial. how should i know the status like busy or free? for now . i know asterisk have QUEUESTATUS variable， QUEUESTATUS The status of the call as a text string, one of TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL but the variable have no busy or free status? how to know the numbers in the queue is busy or not at present? Need your help. thanks! 2010/3/18 ABBAS SHAKEEL shakeel.abbas@gmail.com: Hello, Please have a look to DIALSTATUS variable. here http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS I hope it helps On Thu, Mar 18, 2010 at 1:31 PM, Zhang Shukun bit...@gmail.com wrote: hi,all one problem confuse me these days. i want to sequence dial three PSTN number(a,b,c) first, if i dial number a, if a is busy , i will dial number b. if b is busy, i will dial number c. Dial(SIP/a...@ip,30) Dial(SIP/b...@ip,30) Dial(SIP/c...@ip,30) i want to know before i dial number a, how to know if a is busy now? if a is busy now. i will not dial a, instead, i will dial number b directly. to summary is : in asterisk, how to detect a pstn telephone number is busy or not before dialing it? Thanks! -- Best regards, Sucan -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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 -- Best regards, Sucan -- _ -- 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] confbridge not working?
What does the source code tell you about the circumstances in which
that particular error string is produced?
--
Sent from mobile device
On Mar 18, 2010, at 11:20 PM, Kelvin Chan kelv...@positronics.com
wrote:
Hi guys,
I'm trying to move away from meetme to loose the dependency on dahdi.
ConfBridge seems to be a good fit but I can't get it going. The
document
sounds like an easy to use app. Am I missing any bridge_ modules?
Asterisk 1.6.2.0~rc2-0ubuntu1.2
-- Executing [...@outbound:1] Answer(SIP/109-b877a8c8, ) in
new
stack
-- Executing [...@outbound:2] ConfBridge(SIP/109-b877a8c8,
conf) in new stack
[Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434
join_conference_bridge: Conference bridge '521' could not be created.
dial plan:
exten = _52X,1,Answer()
exten = _52X,n,ConfBridge(${EXTEN})
Thanks,
Kelvin
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Re: [asterisk-users] confbridge not working?
Hi!
Did a quick test, worked as a clock:
exten = 0317998959,1,Set(CHANNEL(language)=se)
exten = 0317998959,n,Answer()
exten = 0317998959,n,ConfBridge(1001,s) 0317998959,n,Hangup()
On Thu, 18 Mar 2010 20:20:35 -0700, Kelvin Chan wrote:
Hi guys,
I'm trying to move away from meetme to loose the dependency on dahdi.
ConfBridge seems to be a good fit but I can't get it going. The document
sounds like an easy to use app. Am I missing any bridge_ modules?
Asterisk 1.6.2.0~rc2-0ubuntu1.2
-- Executing [...@outbound:1] Answer(SIP/109-b877a8c8, ) in new
stack
-- Executing [...@outbound:2] ConfBridge(SIP/109-b877a8c8,
conf) in new stack
[Mar 19 03:16:33] ERROR[2294]: app_confbridge.c:434
join_conference_bridge: Conference bridge '521' could not be created.
dial plan:
exten = _52X,1,Answer()
exten = _52X,n,ConfBridge(${EXTEN})
Thanks,
Kelvin
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Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource
Thanks I will look into it. On Fri, Mar 19, 2010 at 2:26 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: I would like to know if any one have experience with live audio streaming like 1. Streaming from an online resource Look at app_ices and icecast. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Ices 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 -- Best Regards Shakeel Abbas -- _ -- 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
