Re: [asterisk-users] Dial Plan Conf
Hi I want that all of my call should be asked for a code . And then all call should go to a fixed extension. My application will be running there that will differentiate stream of calls. like person A enters 1234 person B enters 2345 both call will be directed to extension say 101, and from there my app will create two audio stream one is by reading code entered by caller . I am currently reading book as instructed. But it would be more helpful if you have already parsed that vdp. On Tue, Oct 26, 2010 at 2:23 AM, Nile Kaledon nile.kale...@gmail.comwrote: Hi, I just downloaded your vdp file and it's working fine on my installation (Asterisk 1.4). Can you be more specific on the issue you experienced? Nile -- _ -- 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] Mobile Phones and Asterisk
Hi, I changed my sip.conf and added call-limit. At first I thought it works ok, since i tried calling a cellphone that is currently busy(phone answers 1st softphone, then another softphone calls the same number, it now returns INUSE). But then, i tried calling a different number while the first phone is busy, but it returns INUSE. It seems that the status being returned was from the peer itself(both phones uses the same peer) and not from the device(mobile phone) which i believe is more logical. I also tried using DIALSTATUS(which of course you need to DIAL first), but then I only hear a busy tone and the dialstatus will return a noanswer. Do I have to configure it first in order to capture the busy status of a device? Have you done something similar to this? I'm using ver. 1.6. Thanks in advance. regards, RYAN ICASIANO From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of GBR Icasiano, Ryan A. [raicasi...@globalbridgeresources.com] Sent: Tuesday, October 26, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Mobile Phones and Asterisk Hi, Is the dev_state can also be used to track a mobile phone's status via SIP? I tried it on several phones(nokia, samsung) but it returns NOANSWER but i can hear a beep beep beep sound indicating that it is currently busy. regards, RYAN ICASIANO -- _ -- 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] Mobile Phones and Asterisk
Dear Asterisk-Users, I have this Asterisk Box I run in my house, I need to terminate and originate remote calls through the box via internet (SIP), the problem is in Nigeria most ISPs would not provide you with Public Addresses, all they provide is dynamic Natted addresses which change each time one connects, I have thought of all possible solutions and cannot come up with one, can anyone please help. Thanks in anticipation ABEJIDE, Ayodele A. (CCNA) +2348039269311 -- _ -- 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] Mobile Phones and Asterisk
Try http://www.dyndns.com/ that should solve your problem with dynamic IPs. Regards, Jonathan On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: Dear Asterisk-Users, I have this Asterisk Box I run in my house, I need to terminate and originate remote calls through the box via internet (SIP), the problem is in Nigeria most ISPs would not provide you with Public Addresses, all they provide is dynamic Natted addresses which change each time one connects, I have thought of all possible solutions and cannot come up with one, can anyone please help. Thanks in anticipation ABEJIDE, Ayodele A. (CCNA) +2348039269311 -- _ -- 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 -- Personal webpage - www.jonbaraq.eu -- _ -- 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] E1 configuration
hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.comwrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com msn%3aflaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- 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 -- _ -- 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 -- free is to know that you have a different option -- _ -- 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] IAX2 call dropped when a second call comes in
Hello list, I have this problem with dropped calls on Asterisk. The setup is SIP internal extensions (Grandstream GXP-2000), two internal analogue DAHDI extensions and IAX2 trunk lines. IAX2 trunks use ulaw/alaw. The Internet connection is ADSL. Asterisk is 1.6.1.6 Everything worked fine until about 1.5 months ago (for 1 year) until the client started to report dropped call. The scenario tends to be: 1. Client is on an external call (through trunk). 2. Another call comes in. 3. As soon as the second call is picked up, first call drops. I thought it might be a bandwidth problem - so I checked upstream and downstream bandwidth. The smallest one is downstream - at about 300kbs. I'm not sure they ever have more then 2 trunk calls at the same time. I have turned logging to verbose in logger.conf, but I just can't see anything that seems relevant in the logs. I can attach the parts of the logs during which I've been told by the client the calls dropped if anyone would like to have a look at them. I can also attach iax.conf or any other config file if you would like to see it. The provider is adamant that there is no problem at their end. Any ideas on this one would be much 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] Mobile Phones and Asterisk
thanks i would check it up ABEJIDE, Ayodele A. (CCNA) +2348039269311 Date: Tue, 26 Oct 2010 12:52:30 +0200 From: jonathan@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Try http://www.dyndns.com/ that should solve your problem with dynamic IPs. Regards, Jonathan On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: Dear Asterisk-Users, I have this Asterisk Box I run in my house, I need to terminate and originate remote calls through the box via internet (SIP), the problem is in Nigeria most ISPs would not provide you with Public Addresses, all they provide is dynamic Natted addresses which change each time one connects, I have thought of all possible solutions and cannot come up with one, can anyone please help. Thanks in anticipation ABEJIDE, Ayodele A. (CCNA) +2348039269311 -- _ -- 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 -- Personal webpage - www.jonbaraq.eu -- _ -- 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] Mobile Phones and Asterisk
Hello Jonathan, The solution would work only if the ISP has one public address, but in my solution they have a pool of public address, any other possible solution? ABEJIDE, Ayodele A. (CCNA) +2348039269311 From: ayodeleabej...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 26 Oct 2010 11:01:09 + Subject: Re: [asterisk-users] Mobile Phones and Asterisk thanks i would check it up ABEJIDE, Ayodele A. (CCNA) +2348039269311 Date: Tue, 26 Oct 2010 12:52:30 +0200 From: jonathan@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Mobile Phones and Asterisk Try http://www.dyndns.com/ that should solve your problem with dynamic IPs. Regards, Jonathan On Tue, Oct 26, 2010 at 12:40 PM, ayodele abejide ayodeleabej...@hotmail.com wrote: Dear Asterisk-Users, I have this Asterisk Box I run in my house, I need to terminate and originate remote calls through the box via internet (SIP), the problem is in Nigeria most ISPs would not provide you with Public Addresses, all they provide is dynamic Natted addresses which change each time one connects, I have thought of all possible solutions and cannot come up with one, can anyone please help. Thanks in anticipation ABEJIDE, Ayodele A. (CCNA) +2348039269311 -- _ -- 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 -- Personal webpage - www.jonbaraq.eu -- _ -- 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
Re: [asterisk-users] E1 configuration
hi, So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well. If you need I can send you such configuration. good look! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Tue, 26 Oct 2010 14:24:13 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- 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 -- _ -- 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 -- free is to know that you have a different option -- _ -- 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
Re: [asterisk-users] 2 HB8 cards in one server - first one is not recognized, the second is
On 10/26/2010 06:38 AM, Administrator TOOTAI wrote: I installed 2 HB8 cards each of them with a Quad Bri modules in a HP 360 G6 running Debian Squeeze. Here is an output of dmesg wafter server has booted: [9.784123] wctdm24xxp :0b:08.0: PCI INT A - GSI 31 (level, low) - IRQ 31 [ 11.847073] bnx2: eth0 NIC Copper Link is Up, 1000 Mbps full duplex [ 11.847600] ADDRCONF(NETDEV_CHANGE): eth0: link becomes ready [ 11.859589] wctdm24xxp :0b:08.0: Timeout waiting for receive frame. [ 14.871333] wctdm24xxp :0b:08.0: Timeout waiting for receive frame. [ 14.871404] wctdm24xxp :0b:08.0: The firmware may be corrupted. Please completely power off your system, power on, and then reload the driver with the 'forceload' module parameter set to 1 to attempt recovery [ 14.893874] wctdm24xxp :0b:08.0: PCI INT A disabled [ 14.893886] wctdm24xxp: probe of :0b:08.0 failed with error -5 [ 14.893911] alloc irq_desc for 30 on node -1 [ 14.893913] alloc kstat_irqs on node -1 [ 14.893919] alloc irq_2_iommu on node -1 [ 14.893927] wctdm24xxp :0e:08.0: PCI INT A - GSI 30 (level, low) - IRQ 30 [ 16.915156] wctdm24xxp :0e:08.0: Timeout waiting for receive frame. [ 17.924645] wctdm24xxp :0e:08.0: firmware: requesting dahdi-fw-hx8.bin [ 17.953971] wctdm24xxp :0e:08.0: Hx8 firmware version: 2.06 snip before asking RMA for the card, I would like to know what you think about this matter. First, Digium technical support would be more than happy I'm sure to help you trouble shoot this. That being said... First thing I would do is update to the current trunk of dahdi-linux. Revision 9397 [1] http://svn.asterisk.org/view/dahdi?view=revisionrevision=9397 was added because of some systems that did not provide reliable polling from the board side, which could result in erroneous your firmware may be corrupted... messages. However, since you have one card that works and one that doesn't I give this a low probability of fixing it. Next, if updating the driver does not help and if the problem follows the card (i.e., you can swap cards and now the second card fails to load), I would disable dahdi from starting automatically, power off your system, remove the working card, power on, and try modprobe wctdm24xxp forceload=1 on the chance that the firmware on the board actually is corrupted. If neither of those things work, you may need to RMA your card. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel Bank ? Simple Switch Hangup?
I am trying to configure a channel bank with 24 ports of FXS., but appear to
be hitting a roadblock? This worked on v1.4.xx but now just get
SimpleSwitch and immediate=no/yes don't seem to make a difference?, no
matter if under top section, under channel, etc.
Chan_dahdi.conf:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A104 port 3 [slot:1 bus:1 span:3] wanpipe3
context=from-cb
group=3
echocancel=yes
signalling=fxo_ls
channel = 49-72
immediate=yes
Extensions.conf:
[from-cb]
exten = s,1,DISA,no-password|internal
[internal]
include = sip-stations
include = iax-trunks
include = outbound
[outbound]
exten = _1XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XXX,1,Dial(DAHDI/g1/${EXTEN})
When I pickup a line, and hit any key I get:
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/72-1'
-- Hungup 'DAHDI/72-1'
Asterisk Version 1.6.2.13
Lastest DAHDI/LibPRI/SpanDSP
--
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Re: [asterisk-users] Channel Bank ? Simple Switch Hangup?
Nevermind, figured it out.
Immediate=yes on top part of chan_dahdi.conf
And in extensions.conf
Exten =s,1,disa(no-password,internal)
William Stillwell
Systems Architect
MDT Personnel, LLC.
Ph. Coming soon.
Fx. Coming soon.
Cl. 727-638-6208
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell (Lists)
Sent: Tuesday, October 26, 2010 8:58 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Channel Bank ? Simple Switch Hangup?
I am trying to configure a channel bank with 24 ports of FXS., but appear to
be hitting a roadblock? This worked on v1.4.xx but now just get
SimpleSwitch and immediate=no/yes don't seem to make a difference?, no
matter if under top section, under channel, etc.
Chan_dahdi.conf:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A104 port 3 [slot:1 bus:1 span:3] wanpipe3
context=from-cb
group=3
echocancel=yes
signalling=fxo_ls
channel = 49-72
immediate=yes
Extensions.conf:
[from-cb]
exten = s,1,DISA,no-password|internal
[internal]
include = sip-stations
include = iax-trunks
include = outbound
[outbound]
exten = _1XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XXX,1,Dial(DAHDI/g1/${EXTEN})
When I pickup a line, and hit any key I get:
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/72-1'
-- Hungup 'DAHDI/72-1'
Asterisk Version 1.6.2.13
Lastest DAHDI/LibPRI/SpanDSP
--
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Re: [asterisk-users] Extension Exists
Thanks Leif, Forgot I could do a db lookup for the ddi. 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
Re: [asterisk-users] Channel Bank ? Simple Switch Hangup?
Have you contacted Sangoma regarding their card configuration?
I have found them always very knowledgeable and helpful
I would certainly go there first.
John Novack
William Stillwell (Lists) wrote:
I am trying to configure a channel bank with 24 ports of FXS., but
appear to be hitting a roadblock? This worked on v1.4.xx but now just
get SimpleSwitch and immediate=no/yes don't seem to make a
difference?, no matter if under top section, under channel, etc.
Chan_dahdi.conf:
[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
;Sangoma A104 port 3 [slot:1 bus:1 span:3] wanpipe3
context=from-cb
group=3
echocancel=yes
signalling=fxo_ls
channel = 49-72
immediate=yes
Extensions.conf:
[from-cb]
exten = s,1,DISA,no-password|internal
[internal]
include = sip-stations
include = iax-trunks
include = outbound
[outbound]
exten = _1XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XX,1,Dial(DAHDI/g1/${EXTEN})
exten = _XXX,1,Dial(DAHDI/g1/${EXTEN})
When I pickup a line, and hit any key I get:
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Starting simple switch on 'DAHDI/72-1'
-- Hungup 'DAHDI/72-1'
Asterisk Version 1.6.2.13
Lastest DAHDI/LibPRI/SpanDSP
--
Dog is my Co-pilot
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[asterisk-users] Auto provisioning from public server
Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -- _ -- 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] Auto provisioning from public server
You can provision over a WAN and access-lists or iptables can limit the networks allowed. Define what level of security you need first. For further security you can use an inbound proxy and check the http headers for agent identification. This can also be faked. Practice layers of security... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Oct 26, 2010 at 12:31 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -- _ -- 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] Auto provisioning from public server
I havent had much auto provisioning experience, however, what about just using IPTables to create an access list essentially for known IPs to connect via HTTP/HTTPS and block all other addresses. This would only work if the phones are coming from a Static IP, but I figured i'd give my 2 cents to try and help. On Tue, Oct 26, 2010 at 11:31 AM, Jonas Kellens jonas.kell...@telenet.bewrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -- _ -- 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 -- Matt -- _ -- 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] Auto provisioning from public server
On Tue, Oct 26, 2010 at 12:31 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Tuesday, October 26, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Auto provisioning from public server You can provision over a WAN and access-lists or iptables can limit the networks allowed. Define what level of security you need first. For further security you can use an inbound proxy and check the http headers for agent identification. This can also be faked. Practice layers of security... ~ Andrew lathama Latham lath...@gmail.com To second Andrew's reply - Auto-provisioning is generally done in a TFTP/HTTP environment. So you will want to set up a layered-vlan environment using IPTABLES or whatever so you can poke freely with constraints. The phone is dumb, so your network needs to be smart... -- _ -- 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] Auto provisioning from public server
On 10/26/2010 05:40 PM, Matt Desbiens wrote: I havent had much auto provisioning experience, however, what about just using IPTables to create an access list essentially for known IPs to connect via HTTP/HTTPS and block all other addresses. This would only work if the phones are coming from a Static IP, but I figured i'd give my 2 cents to try and help. Thank you for your input, but IP-addresses will change, so this would then become an administrative and time-consuming job... Jonas. -- _ -- 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] Auto provisioning from public server
Hello, many SIP phones offer you the possibility to provisioning them over a FTP connection (with username and password). Regards - Bakko -- _ -- 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] Auto provisioning from public server
With the new phones with VPNs you can also do a stepped provision One provisioning service for the vpn and another for the sip that can only be reached with the vpn. This is advanced stuff so take your time and learn about the tech. -- _ -- 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] Auto provisioning from public server
Think about limiting geographically or use a CDN with good controls. Thank you for your input, but IP-addresses will change, so this would then become an administrative and time-consuming job... Jonas. -- _ -- 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] Auto provisioning from public server
On 10/26/2010 05:41 PM, Andrew Latham wrote: You can provision over a WAN and access-lists or iptables can limit the networks allowed. Define what level of security you need first. For further security you can use an inbound proxy and check the http headers for agent identification. This can also be faked. Practice layers of security... Well, what I'm really aiming for is this : I let users make easy config files via web interface. This results in a config file with name MAC-address of the IP-phone. This config file is then available on the public server. User just needs to points his IP-phone to the provisioning URL. Remarks : - User from site A will want other configuration then user from site B. - User from site A may not have access to or download config file of user from site B and vica versa. Expand setup : Also a phone book becomes available from the public server for the users... Jonas. -- _ -- 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] Auto provisioning from public server
On 10/26/2010 05:52 PM, bakko wrote: Hello, many SIP phones offer you the possibility to provisioning them over a FTP connection (with username and password). Regards - Bakko In this case I will want to use Snom phones. TFTP is available, but no FTP (with indeed then a username and password). FTP would be great... Jonas. -- _ -- 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] Auto provisioning from public server
On Tue, Oct 26, 2010 at 12:06 PM, Jonas Kellens jonas.kell...@telenet.be wrote: On 10/26/2010 05:52 PM, bakko wrote: Hello, many SIP phones offer you the possibility to provisioning them over a FTP connection (with username and password). Regards - Bakko In this case I will want to use Snom phones. TFTP is available, but no FTP (with indeed then a username and password). FTP would be great... I wouldn't do this unless your connection is encrypted. -- _ -- 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] Auto provisioning from public server
snom phones can do http digest authentication... In this case I will want to use Snom phones. TFTP is available, but no FTP (with indeed then a username and password). FTP would be great... Jonas. -- _ -- 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] Auto provisioning from public server
Hi! In this case I will want to use Snom phones. TFTP is available, but no FTP (with indeed then a username and password). FTP would be great... You could also consider to use the SNOM Redirection Service for provisioning: http://wiki.snom.com/PROVISIONING Remark: TR-69 provisioning doesn't appear to fit to your environment from what you have disclosed. 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] Auto provisioning from public server
On 26 Oct 2010, at 16:31, Jonas Kellens wrote: has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? What handset? That's rather what controls your options. Some support HTTPS with client certificate authentication. Some support passwords. Some don't. S -- _ -- 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] Trim the RDNIS
What I am needing to do is to trim the 1 from beginning of the RDNIS and I have tried using the CUT function but cannot seem to make it work for me. What we have is a phone number like this, 18881232342 and want to make it like this 8881232342. I appreciate any help that you guys can give. Thanks! -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. crami...@tele-onecom.com 903-531-0777 -- _ -- 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] need to be able to pass a call to the pstn from another pbx trunk
pstn pstn asterisk link between avaya pbx both systems tied together by 2 pri's both have trunks out to the pstn want to get rid of the avaya pstn trunk and send thru my asterisk box avaya still has inbound calls on this trunk until late november (att is dragging their feet doing the porting - 8 weeks between ) and still has stations that we are not in a position to migrate to asterisk just yet (about 500) can get the call to show on the link between systems, but asterisk has no station associate it with so it drops? any suggestions out there? -- _ -- 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] need to be able to pass a call to the pstn fromanother pbx trunk
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Terrell Sent: Tuesday, October 26, 2010 1:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] need to be able to pass a call to the pstn fromanother pbx trunk pstn pstn asterisk link between avaya pbx both systems tied together by 2 pri's both have trunks out to the pstn want to get rid of the avaya pstn trunk and send thru my asterisk box avaya still has inbound calls on this trunk until late november (att is dragging their feet doing the porting - 8 weeks between ) and still has stations that we are not in a position to migrate to asterisk just yet (about 500) can get the call to show on the link between systems, but asterisk has no station associate it with so it drops? any suggestions out there? Dump the avaya calls to a local channel or conference? -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc., for outbound calls, it acts basically like a fancy click-to-call application. So... You need Asterisk to login into GV, and initiate the call. GV will dial the number you tell it to, then connect it to one of your GV numbers. In my case, the AGI is what connects to GV and initiates the call. GV, then dials the number I told it to dial, then connects it with my ipKall number (which I have as one of my GV numbers). In Asterisk, the outbound call runs the AGI and places the channel in the DB, then waits for an incoming call via my inbound ipKall trunk. Once the ipKall comes into Asterisk, the Bridge command is used to bridge the original (with the matching DB entry) call-- the call that is coming in from GV through ipKall. I suppose you don't need that AGI and could probably do this using Curl in the dialplan. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E That makes sense but I do not see where the new feature is in Asterisk 1.8 which include Google Voice support per http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt: 290973 |dvossel|Make outbound Google Voice calls. | | It seems that the GV has been a feature for sometime with previous versions? I'm just trying to keep the process as simple as possible and seeing three different methods is a little confusing: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ (no script referenced) http://www.davidvossel.com/?p=28 (python script and listed in the change log above) http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script) Is your .agi and .git the same script? I do not have a git client on this host to see for myself. Thanks, Stephen -- _ -- 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 media being sent in SIP call
Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone calling on a different trunk works just fine. Any ideas? -- Take care and have fun, Mike Diehl. -- _ -- 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] Trim the RDNIS
On Tue, 26 Oct 2010, Chris Ramirez wrote: What I am needing to do is to trim the 1 from beginning of the RDNIS and I have tried using the CUT function but cannot seem to make it work for me. What we have is a phone number like this, 18881232342 and want to make it like this 8881232342. I appreciate any help that you guys can give. Thanks! Read whereever-you-keep-your-asterisk-sources/doc/README.variables. -- 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] CALLERPRES() with Queue
Hi, Bump to see if anyone can help us too. Really this is a problem. I don't want to show the caller id number and name to the Agent in certain conditions. Changing the CID will mess the CDR/Queue log and this is not the acceptable behavior. In the Dial app, everything is OK. Alexandre Em 06-10-2010 17:35, Rodrigo Lang escreveu: Good afternoon list, I'm having a problem using the function CALLERPRES() when connection to a Queue(). When I call an extension, before the Dial (), I select the function CALLERPRES () as unavailable to link the extension comes as anonymous. But if I call a queue before the Queue (), I select the function CALLERPRES() as unavailable, but the identification appears normal. Is it a problem or configuration? Someone can have for that? Regards, -- Rodrigo Lang http://rodrigorecipes.blogspot.com/ http://rodrigorecipes.blogspot.com/2010/08/ssh-rapido-e-pratico.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: SMS inbound
Hi guys, a little OT but I figured this is the place that would know. Is there a free or paid webapp where I can get inbound sms messages? I only need to receive a few inbound sms messages a month but it cant be my current cell number :-( Any thoughts? Cheers, Dean -- _ -- 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 media being sent in SIP call
2010/10/26 Mike Diehl mdi...@diehlnet.com Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone calling on a different trunk works just fine. Any ideas? codec incompatibilities ? t.38 ? -- Take care and have fun, Mike Diehl. -- _ -- 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] OT: SMS inbound
Google voice... ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Oct 26, 2010 at 4:41 PM, Dean Collins d...@cognation.net wrote: Hi guys, a little OT but I figured this is the place that would know. Is there a free or paid webapp where I can get inbound sms messages? I only need to receive a few inbound sms messages a month but it cant be my current cell number L Any thoughts? Cheers, Dean -- _ -- 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] E1 configuration
dear please send these configurations. thanks On Tue, Oct 26, 2010 at 3:04 PM, Flavio Miranda flaviormira...@hotmail.comwrote: hi, So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well. If you need I can send you such configuration. good look! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com msn%3aflaviormira...@hotmail.com Skype: flaviormiranda -- Date: Tue, 26 Oct 2010 14:24:13 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- 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 -- _ -- 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 -- free is to know that you have a different option -- _ -- 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] E1 configuration
Hi, /etc/dahdi/system.conf Att,# Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/ span=1,1,0,cas,hdb3cas=1-15:1101dchan=16cas=17-31:1101#echocanceller=mg2,1-15,17-31 /etc/asterisk/chan_dahdi.conf [trunkgroups] [channels] usecallerid=yescallwaiting=yesusecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yessignalling=mfcr2mfcr2_variant=brmfcr2_get_ani_first=nomfcr2_max_ani=20mfcr2_max_dnis=4mfcr2_category=national_subscribermfcr2_logdir=span1mfcr2_call_files=yesmfcr2_logging=allmfcr2_mfback_timeout=-1mfcr2_metering_pulse_timeout=-1mfcr2_allow_collect_calls=yesmfcr2_double_answer=nomfcr2_immediate_accept=yesmfcr2_forced_release=nomfcr2_charge_calls=yes;language=pt_BRcontext=Saida-de-ligacoesgroup=0callgroup=0pickupgroup=0channel = 1-15,17-31immediate=no#include dahdi-channels.conf /etc/asterisk/dahdi-channels.conf ; Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 (MASTER) HDB3/group=0,11context=Saida-de-ligacoesswitchtype = nationalsignalling = pri_cpechannel = 1-15,17-31context = defaultgroup = 63 Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Wed, 27 Oct 2010 00:15:01 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration dear please send these configurations. thanks On Tue, Oct 26, 2010 at 3:04 PM, Flavio Miranda flaviormira...@hotmail.com wrote: hi, So, I think it depend of what environment are you setting up your link . In my case, E1 R2 Digital Brazil standard (Variant=br), I needed to change dahdi-channels parameter,chan_dahdi.conf , system.conf as well. If you need I can send you such configuration. good look! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Tue, 26 Oct 2010 14:24:13 +0330 From: seighal...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration hi my friend would ou say what did you do for solving the problem? because i use a digium te121p and have many problems. thanks in advance On Mon, Oct 25, 2010 at 4:50 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Sorry, thats right!! I the nest email I will post here what I did in order to sove my problem! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 24 Oct 2010 23:59:27 -0700 From: shakeel.abbas@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E1 configuration although I don't need the solution personally But would like to request you that instead of posting forget it . if you post the solution to the problem it will be more helpful. In case some one else faces the same problem he can use your solution Good luck On Sun, Oct 24, 2010 at 7:10 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Forget it !! After several attempts, I have solved !!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda From: flaviormira...@hotmail.com To: asterisk-users@lists.digium.com Date: Sun, 24 Oct 2010 22:28:16 -0200 Subject: [asterisk-users] E1 configuration Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341...@local:1] Dial(SIP/4804-, DAHDI/g11/21341400,,t) in new stack == Everyone is busy/congested at this time (1:0/0/1) == Spawn extension (local, 21341400, 2) exited non-zero on 'SIP/4804-' The boad has succesfully installed: Digium Wildcard TE110P T1/E1 Card 0 OK 0 0 0 CAS HDB3 0 db (CSU)/0-133 feet (DSX-1) the channels are correct and mfcr2 too, but the calls dont go out. Thanks for any help. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 -- Best Regards Shakeel Abbas --
Re: [asterisk-users] Asterisk 1.8 IAX Registration
On Sun, Oct 24, 2010 at 2:20 PM, Nic Colledge n...@njcolledge.net wrote: I made a debug log of the register and unregister process for a single Zoiper client using IAX and have emailed it direct to you. The error shows in the file as: [Oct 24 19:07:32] ERROR[1403] netsock2.c: getnameinfo(): ai_family not supported I'm going to try and look at this during Astricon :) -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) | Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger -- _ -- 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
