Re: [asterisk-users] Tell apart between network disruption and asterisk restart via AMI
From AMI you can get uptime. If the uptime is short likely Asterisk restarted. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Oct 19, 2012, at 10:31 AM, Alex Villacís Lasso wrote: I have a program that connects to the Asterisk Manager Interface through port 5038 on a remote machine. Suppose I get a TCP disconnection on my program. The program will then attempt to reconnect to the AMI and will eventually succeed. Is there a way to check whether the disconnection was caused by a network disruption, or an Astersk restart/crash? In other words, is the Asterisk process I contacted now the same as the one I was connected before, or is it a different one? The reason I want to know is that I have a cache of information that is costly to parse (scales linearly with the number of extensions) and I want to know how to realize that the information is now stale. -- _ -- 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] Use of Sangoma D500
Does anyone on the list have any experience with using a Sangoma D500 card with Asterisk to transcode G729? If you could mention pros and cons I would like to hear opinions. Thanks -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Authentication: username and password, also to be from the LAN
Is that now permit and deny are used for. To specify the acceptable IP address(es) the user can connect from? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 26, 2012, at 10:11 AM, bilal ghayyad wrote: Hi All; Is it possible to restrict the authentication to be based on the username and password and to be allowed for IPs within the LAN (for example, 192.168.10.x)? I do not need it to be based on the IP only and do not need it to be based on the username and password only, but I need it to be based on the username password and to be from the specific range, so if the IP address of the client was of the range 192.168.10.x then it is allowede to register with its username and password. No need to specify the IP. If it possible, then is it possible to be a configuration per user? Regards Bilal -- _ -- 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] Finish ChanSpy() when channel spied hangs up
I had submitted a patch some time ago to add option s to chanspy. This would
cause chanspy to exit once the specified change was not longer there. I do not
know if it ever got into a released version as I use ABE. It was not in 1.6 but
might be in 1.8.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 8, 2012, at 4:20 AM, equis software wrote:
I need call to C every time that A call to B, but when A-B hangs up i need to
hang up Asterisk-C call too.
Anyboby know another solution?
On Wed, Mar 7, 2012 at 2:51 PM, equis software equissoftw...@gmail.com
wrote:
Here's my dialplan...
[default]
exten = _X.,1,System(echo -e Channel: SIP/519912@SOFTSWITCH\\nContext:
spy\\nExtension: 23\\nSet:SPYCHANNEL=${CHANNEL} /tmp/${UNIQUEID}.call)
exten = _X.,n,System(mv /tmp/${UNIQUEID}.call /var/spool/asterisk/outgoing/)
exten = _X.,n,Dial(SIP/${EXTEN}@SOFTSWITCH)
[spy]
exten = s,1,Answer
exten = s,2,Chanspy(${SPYCHANNEL}|q)
exten = s,3,Hangup
A call to B
and C (519912) is called by Asterisk to spy the call.
Whe the A-B conversation over, C continue connected to Asterisk, I need
Asterisk hangs up this call.
In my case C is another machine that records the call and can´t hang up when
A-B has finished because it doesn't know.
I don't know if i'm clear
On Wed, Mar 7, 2012 at 1:12 PM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Doesn't this automatically finish ?
Jonas.
On 03/07/2012 05:03 PM, equis software wrote:
Is there any way to do this?
Thanks
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Re: [asterisk-users] Problems with codec translation when using Monitor and MixMonitor
Here is a matrix we put together about g729 license needs: == = == === Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln defined record monitor encoders decoders == = == === ulaw ulaw yes yesyes00 ulaw ulaw yes yesno 00 ulaw ulaw yes no no 00 ulaw ulaw yes no yes00 ulaw ulaw no yesyes00 ulaw ulaw no yesno 00 ulaw ulaw no no no 00 ulaw ulaw no no yes00 ulaw g729 yes yesyes33 ulaw g729 yes yesno 23 ulaw g729 yes no no 11 ulaw g729 yes no yes33 ulaw g729 no yesyes33 ulaw g729 no yesno 23 ulaw g729 no no no 11 ulaw g729 no no yes33 g729 ulaw yes yesyes25 g729 ulaw yes yesno 25 g729 ulaw yes no no 11 g729 ulaw yes no yes23 g729 ulaw no yesyes25 g729 ulaw no yesno 25 g729 ulaw no no no 11 g729 ulaw no no yes23 g729 g729 yes yesyes47 g729 g729 yes yesno 37 g729 g729 yes no no 11 g729 g729 yes no yes45 g729 g729 no yesyes47 g729 g729 no yesno 37 g729 g729 no no no 11 g729 g729 no no yes45 -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote: On 01/12/2012 11:57 AM, Daniel - Asterisk wrote: The simplest answer, I purchased one additional license and one simultaneous call is being recorded now. I do not understand why the ulaw codec (or format) is involved here (... No translator path from alaw to unknown ...) Any entry will be very
Re: [asterisk-users] Attempt to Originate between IAX2/xxxx and an application hangs until timeout in 1.8.8.1
One good thing is now that you know what the problem is you should be able to work with zopier support and get them to fix zopier. They have been very responsive to a couple problems I have and I am running the free version. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2012, at 3:03 PM, Kevin P. Fleming wrote: On 01/12/2012 11:58 AM, Alex Villacís Lasso wrote: I have discovered the root cause of the issue. Due to a peculiarity of Zoiper 2.18, this program will *not* send a ACCEPT or RINGING packet back to Asterisk unless the NEW packet that announces the incoming call contains an IAX_IE_CALLING_NUMBER information element. It does not matter if the calling number is empty, but the corresponding IE must exist. This behavior is a change between Asterisk 1.6 and Asterisk 1.8. Well, I applaud your troubleshooting skills and analysis... well done! Unfortunately, that IE is *not* mandatory in an IAX2 NEW packet, and thus Zoiper failing to properly process such NEW packets is a bug in Zoiper. Yes, Asterisk's behavior has changed (since Caller ID handling was overhauled in Asterisk 1.8, while adding Connected ID support), but both the old and new behavior are compliant with the IAX2 protocol. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- _ -- 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] Answering call from queue, then put back in queue?
One way to deal with this is to have two queues. Give priority to the original queue callers land in. Once answered put the call in to the second queue. They will then be in the second queue in the order the agents answered the first queue. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 8, 2012, at 10:26 AM, Todd Routhier wrote: Version: Asterisk 1.8.x Question: Is it possible for an agent to answer a call from a queue, then place the call back in the queue in the same position they were in? Seems that the answer would be yes to the remove from queue, then place back in by having the agent just transfer the call back to the queue but is there any way to put them back in line where they were? The idea is that the owner of the queue doesn't want callers waiting on hold without first having an agent at least answer the call and ask them to please hold. What's the best way to handle this? Thanks in advance! --Todd -- _ -- 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] calling specific 1800-number not going through.
It took 36 seconds for that number to answer when I called it and it looks like
the call hung up after 32000 ms when you dialed via asterisk.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 5, 2012, at 5:45 PM, Joseph wrote:
I have a strange problem.
I'm using the same dialplan to call 1800-number:
[toll-free]
;second 7 audiocodes strips
exten = _71800XXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr)
When I call this number (through pstn-5665) 18005000347 the phone always
rings busy.
When I call any other 1800-number the calls goes through.
When I call the same phone number 18005000347 through a different line the
calls goes through every time.
Here is call (busy) trace to that 18005000347 with sip debug ON:
Can anybody decipher why I'm getting busy signal to that particular
1800-number but not others?
--- SIP read from UDP:10.0.0.110:5060 ---
OPTIONS sip:gateway@10.0.0.110 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
Max-Forwards: 70
From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
To: sip:gateway@10.0.0.110
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Contact: sip:gateway@10.0.0.110:5060
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0
-
--- (12 headers 0 lines) ---
Looking for gateway in default (domain 10.0.0.110)
--- Transmitting (NAT) to 10.0.0.110:5060 ---
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060
From: sip:gateway@10.0.0.110:5060;tag=1c1457828994
To: sip:gateway@10.0.0.110;tag=as7091ae01
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0
Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in
32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 81.15.150.20:5060:
OPTIONS sip:sip.actio.pl SIP/2.0
Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5
Max-Forwards: 70
From: asterisk sip:asterisk@10.0.0.100;tag=as64f6417c
To: sip:sip.actio.pl
Contact: sip:asterisk@10.0.0.100:5060
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
User-Agent: Centrala
Date: Fri, 06 Jan 2012 01:39:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Length: 0
---
--- SIP read from UDP:81.15.150.20:5060 ---
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP
10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715
To: sip:sip.actio.pl;tag=4fc8ac12
From: asterisksip:asterisk@10.0.0.100;tag=as64f6417c
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
Content-Length: 0
-
--- (7 headers 0 lines) ---
Really destroying SIP dialog
'66070317301f64861df62d20769ba385@10.0.0.100:5060' Method: OPTIONS
-- Accepted AUTHENTICATED TBD call from 10.0.0.108
--- SIP read from UDP:10.0.0.110:5060 ---
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360
Max-Forwards: 70
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Contact: sip:11@10.0.0.110:5060;expires=3600
Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Length: 0
-
--- (12 headers 0 lines) ---
Sending to 10.0.0.110:5060 (NAT)
--- Transmitting (no NAT) to 10.0.0.110:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110;tag=as21c548bd
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=3a451a5b
Content-Length: 0
Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000
ms (Method: REGISTER)
--- SIP read from UDP:10.0.0.110:5060 ---
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428
Max-Forwards: 70
From: sip:11@10.0.0.110;tag=1c1472330741
To: sip:11@10.0.0.110
Call-ID: 809487713120129287@10.0.0.110
CSeq: 246 REGISTER
Authorization: Digest
username=11,realm=asterisk,nonce=3a451a5b,uri=sip:10.0.0.100,algorithm=MD5,response=5dd6df18064f3d23cb86ca306820e596
Contact: sip:11@10.0.0.110:5060;expires=3600
Allow:
REGISTER,OPTIONS
Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that?
If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: [general] dateformat=%F %T [logfiles] full = notice,warning,error,debug,verbose,dtmf,fax However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
Yes, you are missing the fact that the verbose setting controls what level of output will be generated in the first place. You can raise and lower the amount of stuff logged/printed on CLI. The lines in logger.conf control what types of lines go to which place. One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:24 PM, Bruce B wrote: Okay, but I thought that the line console = is supposed to be for CLI and the line Full = is supposed to be for the file /var/log/asterisk/full. Why would the Full = be effected by core set verbose 0? Is this just bad assumption on the part of the developers? I would only assume that core set verbose 0 should only effect what I see at CLI level and not at my my /var/log/asterisk/full log file. Am I missing something? Thanks for the feedback. On Fri, Dec 30, 2011 at 6:19 PM, Jim Dickenson dicken...@cfmc.com wrote: If you want to stop stuff from going to the console you can use the command logger mute and console will not get output but log file will. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 3:11 PM, Bruce B wrote: Hi everyone, I am playing around with Asterisk 1.8.8.0 from Digium repository. This is all there is to my logger.conf file: [general] dateformat=%F %T [logfiles] full = notice,warning,error,debug,verbose,dtmf,fax However, when I do, core set verbose 0 at CLI, Asterisk ceases to write to /var/log/asterisk/full file for some reason. When I type core set verbose 9 at CLI then it starts writing to /var/log/asterisk/full. Is this the correct behaviour or am I missing a config setting? Of course I want the /var/log/asterisk/full file to always keep the logs regardless of what the verbosity at CLI level is. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] High verbose set at console effects the logger file Full - Why is that?
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 30, 2011, at 4:55 PM, Bruce B wrote: One can set the verbose level as well as the debug level. These control how much log information is generated at all not where it is being written. What do you mean by above? Can I see something in the logger.conf that will keep it always at certain verbose level regardless of what command I issue at CLI? No the verbose command controls how much verbose stuff is output. The debug command controls how much debug stuff is output. These are absolute controls of that information. As I said in my original email you can turn off stuff going to the CLI with the logger mute command. That way you do not adjust the verbose level at all. You see the problem I have is that Fail2ban reads the asterisk full log file. So, if I am playing on the CLI and then do core set verbose 0 and exit the box and forget to set it back to 9 then Fail2ban stops working because the log file hasn't logged the attack. I still think there is a way around this and I am missing a config. Why would anyone tie security logs to a mere CLI command? Thanks again -- _ -- 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] how to listen on different sip port for a device?
Why not use IAX trunk instead of SIP. This would make it very easy to talk between the two * systems. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 26, 2011, at 4:07 PM, sean darcy wrote: On 12/26/2011 05:43 PM, Yaroslav Panych wrote: 2011/12/26 sean darcyseandar...@gmail.com: So how do I get * to listen to two different ports? sip.conf section [general] bindport=whatever-port-you-want Thanks, but the problem is to get more than 1 port, 5060 and (at least) one other. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] GOIP GSM to SIP Gateway?
I would think it would be better to set a variable for each user and then have
a single context with something like:
_NXX,1,Dial(SIP/${WhatToUse}/${EXTEN})
Or something like this.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 20, 2011, at 1:03 PM, John Kiniston wrote:
On Tue, Dec 20, 2011 at 12:39 PM, Matt mhop...@gmail.com wrote:
Is there anyway (short of defining dial an 8 from this phone for this
trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
get it to use certain SIM cards when calls are made from certain
phones?
You could define multiple contexts with different pattern matches for each
GSM connection and and set your phones to use them, phones 1-3 in context1,
phones 4-6 in context2, etc.
[context1]
_NXX,1,Dial(SIP/GSM1/${EXTEN})
[context2]
_NXX,1,Dial(SIP/GSM2/${EXTEN})
[context3]
_NXX,1,Dial(SIP/GSM3/${EXTEN})
--
A human being should be able to change a diaper, plan an invasion, butcher a
hog, conn a ship, design a building, write a sonnet, balance accounts, build
a wall, set a bone, comfort the dying, take orders, give orders, cooperate,
act alone, solve equations, analyze a new problem, pitch manure, program a
computer, cook a tasty meal, fight efficiently, die gallantly. Specialization
is for insects.
---Heinlein
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Re: [asterisk-users] Play audio file for both Caller and Callee in a call
You also use AMI to inject audio into the conversation using the ChanSpy application. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote: You can’t per se, but you can call an AGI using stream? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of c.savinov...@itntelecom.com Sent: Thursday, December 15, 2011 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call Dear Danny: How can you use Playback in the middle of 2 channels engaged in a conversation? Thanks C. Savinovich Original Message Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in a call From: Danny Nicholas da...@debsinc.com Date: Thu, December 15, 2011 9:31 am To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Playback? What flavor of Asterisk are you using? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS ARNAL Sent: Thursday, December 15, 2011 10:29 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Play audio file for both Caller and Callee in a call Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don’t want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra política de envío y recepción de correo electrónico en el enlace situado más abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at. http://www.tid.es/ES/PAGINAS/disclaimer.aspx -- _ -- 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] Play audio file for both Caller and Callee in a call
Use an AMI packet like this:
Action: Originate
Channel: Local/do_playback@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280
ActionID: PlayBack
Async: true
With dialplan like this:
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
You need to issue an AMI packet for each leg of the call. Each leg will hear
the same audio feed offset by however long it takes the packets to be
processed. In general this is a few milliseconds and should not be a big deal.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 10:27 PM, virendra bhati wrote:
Hi,
Plese give a little example of script so that it will be clear.
On Thu, Dec 15, 2011 at 11:09 PM, Jim Dickenson dicken...@cfmc.com wrote:
You also use AMI to inject audio into the conversation using the ChanSpy
application.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Dec 15, 2011, at 9:23 AM, Danny Nicholas wrote:
You can’t per se, but you can call an AGI using stream?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
c.savinov...@itntelecom.com
Sent: Thursday, December 15, 2011 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Play audio file for both Caller and Callee in
a call
Dear Danny:
How can you use Playback in the middle of 2 channels engaged in a
conversation?
Thanks
C. Savinovich
Original Message
Subject: Re: [asterisk-users] Play audio file for both Caller and
Callee in a call
From: Danny Nicholas da...@debsinc.com
Date: Thu, December 15, 2011 9:31 am
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Playback? What flavor of Asterisk are you using?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ISABEL ORDAS
ARNAL
Sent: Thursday, December 15, 2011 10:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Play audio file for both Caller and Callee in a
call
Dear all,
Anyone of you knows how to play an audio file at the beginning of a call for
both Caller and Callee?
A(x) of Dial application only plays audio for callee. I don’t want to use
MeetMe because I want to use Monitor and MixMonitor.
Thank you!
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
nuestra política de envío y recepción de correo electrónico en el enlace
situado más abajo.
This message is intended exclusively for its addressee. We only send and
receive email on the basis of the terms set out at.
http://www.tid.es/ES/PAGINAS/disclaimer.aspx
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Thanks and regards
Virendra Bhati
+91-8885268942
Software Engineer
Re: [asterisk-users] AMI: anything to glue originate to events?
The easiest thing to do is to create userevents in your dialplan to passed to AMI details you want to key off of. In the original originate you can set so variable that you pass to various macros and what have you. These then generate userevents that AMI can use to track the flow of the call. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 17, 2011, at 12:02 PM, giovanni.v wrote: On 17/11/2011 19.45, c.savinov...@itntelecom.com wrote: if it is what I think it is, I remember I had a similar situation a few years ago, and I ended up having to create an internal table in my code, so that I could keep track of the channel ids + action ids . Which is exactly what I'm doing but I tried to figure out if there was something more reliable ... I refer to the logic not the data structure. Ignoring for a moment the relationship between events, my conclusion is still that there is nothing that ensures that very first event that I will receive after /Response/ to my /Originate/ for that channel is really fired from my application, I can only guess it is. 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 -- _ -- 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] 2 pbxes
Yes. If you have two asterisk boxes running you can trunk them together and place calls from one to to the other. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 3, 2011, at 11:36 AM, mattias wrote: if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -- _ -- 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] permit -- deny not working
I do not know if order is important but I always deny all then permit what I want to permit. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Oct 11, 2011, at 1:15 PM, hussein korbani wrote: Hello, i am having an issue with the DENY permit thingy in the Extensions.conf whenever i use the permit deny , all the calls coming from another sip-trunk to my asterisk ,start to fail doesn't use the Extensions dial plans that i created my context contain the following: [context1] . host=1.2.3.4 permit=1.2.3.4/255.255.255.255 deny=0.0.0.0/0.0.0.0 .. can anyone explain why its failing? P.S; calls from extension created on the ASterisk works fine even when permit\deny is used or not Best regards, Hussein K -- _ -- 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] Add SIP diversion header in originate from AMI?
You can dial a local channel which executes a dial plan that does what you want.
Channel: Local/dial_number@cfmc_cdi_private
This will use exten dial_number in the cfmc_cdi_private context.
If you add something like this to the originate packet
Variable: CfMC_Use_CID=5419712513
You can use ${CfMC_Use_CID} to get the value.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Oct 7, 2011, at 8:03 AM, Tobias Steen wrote:
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected to
an external customer service hired by another company.
My question:
How can I add this header in a originateaction call via AMI?
Does the originated calls go through any context where I can add this header
with dialplan functions like AddSipHeader() or is it possible to dothis
directly in the OriginateAction through AMI?
Example from voip-info:
[macro-diversion-header]
exten = s,1,SIPAddHeader(Diversion:
tel:+{ARG1}\;reason=user=busy\;screen=no\;privacy=off)
Best regards
Tobias
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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls
I do not know when the recording actually starts but if it start when the agent
answers the call then it might be possible to have the name set in an AGI that
gets run when the agent answers call. If nothing else you can set a variable to
the name you want to have the file have and rename it at end of call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 27, 2011, at 10:30 PM, Sam Govind wrote:
:P I'd this very similar situation/ project Carl - and guess what. The
filename is created before the call actually hits QUEUE application so these
Queue variables are not populated by then so filename won't contain the Agent
Number.
UNLESS you move the file after queue to a new filename containing the Agent
Number.
like ;
exten = whatever,n,SET(MONITOR_FILENAME=blah-blah)
exten = whatever,n,Queue(${params}); Queue should contain option c to
continue in dialplan when callee hangup. Caller hangup case needs special
attention too
exten = whatever,n,System(mv ${old-Filename}
${old-Filename}-${MEMBERINTERFACE})
I guess this should do the job.
On Tue, Sep 27, 2011 at 8:30 PM, Carlos Chavez cur...@telecomabmex.com
wrote:
On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
Dears;
I am facing now a problem in the recording the calls that coming via the
queue, the problem that I am not able to make the filename contains the
agent (for example its extension) who received the call.
Actually by looking to the below settings, it is clear that the agent name
(it the phone extension or it is sip username .. etc) will not be included
in the filename.
How can I include the agent name in the filename? Because in outboud it is
easy as the ${CHANNEL} will contain the sip username of the IP Phone but in
the outbound it will contain the DAHDI channel that the call came via it ..
so How to inlude the sip username for the IP Phone of the agent that is
going to get the call from the queue?
exten =
s,1,Set(MONITOR_FILENAME=${CHANNEL}${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
exten = s,2,Queue(OrangeCMG,t,,,180)
exten = s,3,Macro(voicemail,SIP/reception)
Regards
Bilal
; If set to yes, just prior to the caller being bridged with a queue
member
; the following variables will be set
; MEMBERINTERFACE is the interface name (eg. Agent/1234)
; MEMBERNAME is the member name (eg. Joe Soap)
; MEMBERCALLS is the number of calls that interface has taken,
; MEMBERLASTCALL is the last time the member took a call.
; MEMBERPENALTY is the penalty of the member
; MEMBERDYNAMIC indicates if a member is dynamic or not
; MEMBERREALTIME indicates if a member is realtime or not
;
;setinterfacevar=no
Basically the variable ${MEMBERINTERFACE} will have the extension (if
using dynamic members) or the agent number.
--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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Re: [asterisk-users] Question about Registrations
One way of doing something when a peer registers is to use AMI to monitor events and when a register event occurs do what you want. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 23, 2011, at 7:08 PM, Alex Balashov wrote: On 09/23/2011 09:59 PM, CDR wrote: In Trunk, or earlier, is it possible to execute an AGI or any piece of the Diaplan when a new peer registers successfully? No. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.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 -- _ -- 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] Connecting to a Taqua switch
My provider has always sent the SIP control info from one IP and the media packets from another. As long as your firewall passes the data there should be no problem. I did not have to do anything special in my configuration. This is using ABE which is based on 1.4. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 22, 2011, at 10:09 AM, Philip Prindeville wrote: Anyone have any configuration experience connecting Asterisk 1.8 to the PSTN via SIP on a Taqua 7000 switch? My local carrier recently upgraded software and changed their configs so that signalling and media are on different cards (and hence different IP addresses), and it's causing issues. I suspect there are other factors at play... it may or may not be behind a properly configured SBC. Thanks, -Philip -- _ -- 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] DTMF issues still
I had a very strange problem with a Sangoma card that I had both Sangoma (about 3 hours) and Digium (about 2 hours) look at. When I got a different Sangoma tech to look at the problem it went away. I told the tech he did something and he said I alway verify the firmware on the card is updated and as it was not I updated it. That fixed the problem. This system had worked before a dahdi update was applied. Bottom line make sure you have the most current firmware for your card. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 8, 2011, at 2:49 PM, vmed...@apcn.net wrote: I am still having major issues with dtmf recognition. My setup is Polycom end points. Tried this with different models, firmware and cfgs. Outbound calls are not going out reliably. Phones are set to rfc2833. I have had sangoma and elastix support look at it.. No better. Running asterisk 1.8.4. What am I missing? Any help appreciated.. Btw.. Used a basic test set and dialed out on all lines no problem. Sangoma card is a a400 with echo cancel. Sent from my android device. -- _ -- 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] Agent Login, Logout, Ready, Not Ready from the CTI application
You need to use the AMI interface an deal with the events that are give to you. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 4, 2011, at 4:36 PM, bilal ghayyad wrote: Hi All; We know that agents can login and logout from the phone handset. But if we need the login, logout, ready and not ready to be from an application and to be integrated with the CRM, how to acheive this? Normally in Cisco and AVAYA, they use CTI integration and the CTI client (which is embded in the CRM application) will receive the the caller id or information via that CTI client. How this to be done in Asterisk? By the way: is the ready and not ready in Asterisk? Regards Bilal -- _ -- 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] chanspy spies on wrong channel
The argument to chanspy is a pattern and not an exact match. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 2, 2011, at 3:48 PM, steve casto wrote: asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel 2.0 (from extensions.conf) exten= 304,1,ChanSpy(Zap/4|q) exten= 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304@flash:1] ChanSpy(Zap/31-1, Zap/4|q) in new stack [Jul 1 18:24:47] VERBOSE[14447] logger.c: == Spying on channel Zap/41-1 [Jul 1 18:24:47] NOTICE[14447] app_chanspy.c: Attaching Zap/31-1 to Zap/41-1 If while spying on Zap/41 that channel is hung up: [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Done Spying on channel Zap/41-1 [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Spying on channel Zap/4-1 [Jul 1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Zap/31-1 to Zap/4-1 thanks list Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] background audio for inbound leg
The way I play a sound file into a bridged call is to use chanspy w option. I do this with an application that does AMI commands. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 17, 2011, at 10:25 AM, Tom Browning wrote: Is there an easy way to feed an audio file (think background music, ever so softly) to the inbound leg of a bridged call (and not send / mix it to the outbound leg)? exten = blah,1,Answer() exten = blah,2,StartSomeAudio(foo)? exten = blah,3,Dial(SIP/bar) Where the audio would continue to play to the inbound leg in addtion to the bridged inbound audio. Thanks in advance including any RTFM references :-) Tom -- _ -- 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] issues.asterisk.org/jira not working
I get this on my Mac: Safari can’t open the page. Safari can’t open the page “https://issues.asterisk.org/jira/browse/ASTERISK-17984” because Safari can’t establish a secure connection to the server “issues.asterisk.org”. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 11:38 AM, William Stillwell wrote: You mean this one? https://issues.asterisk.org/jira/browse/ASTERISK-17984 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, June 08, 2011 2:17 PM To: asterisk-users Subject: [asterisk-users] issues.asterisk.org/jira not working Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!! -- _ -- 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] issues.asterisk.org/jira not working
If I click on the link below, without jira, Safari goes to here: https://issues.asterisk.org/main_page.php And yes it works. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 8, 2011, at 1:54 PM, Kevin P. Fleming wrote: On 06/08/2011 02:27 PM, Andrew Latham wrote: On Wed, Jun 8, 2011 at 3:20 PM, Russell Bryantruss...@digium.com wrote: A number of people are reporting that Safari is not working properly with JIRA. Use Firefox or Chrome for now. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org This could be an issue with the CA keys used in Safari. I remember having to chain load a root key for a server just for iphone support a while back. looking Apache option is SSLCertificateChainFile /full/path/to/your.ca-bundle Can Safari open a connection to https://issues.asterisk.org? (no /jira suffix) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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 -- _ -- 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] Trying out a new version with sangoma card
In asterisk CLI do pri show spans. The fact the card is in RED alert means the hardware does not see the pri line connected to the card. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 11, 2011, at 6:55 PM, Nicolas Ross wrote: Le 2011-05-09 09:31, Jim Dickenson a écrit : Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. It appears it did not change anything... So, to re-cap, I have a sangoma A101 card, with the firmware uptodate, on the asterisk 1.8.3.3, dahdi-linux 2.4.1.2, and dahdi-tools 2.4.1. When asterisk is running, cat /proc/dahdi/1 yields : Span 1: WPT1/0 wanpipe1 card 0 (MASTER) B8ZS/ESF RED 1 WPT1/0/1 Clear (In use) 2 WPT1/0/2 Clear (In use) (...) 24 WPT1/0/24 Hardware-assisted HDLC (In use) And when it's not, the (In use) go away. When, dialing I get Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) So, does anybody got any idea ? -- _ -- 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] Trying out a new version with sangoma card
Make sure the firmware on the card is latest. I had a problem, not like your, and flashing the card to the latest firmware resolved it. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 9, 2011, at 6:11 AM, Nicolas Ross wrote: Hi ! We curently have a centos 5 / asterisk 1.4 server that we have some DTMF problems with. It has a Sangoma A104d card and only port one is used to connect to the PSTN. Port 2 is conencted via a cross-over cable to a RAS for modem access and port 3 is connected for data communication via PPP. Now, I want to freshen this setup to something newer. So I installed a Scientific Linux 6 server, with asterisk 1.8 and the latest Sangoma drivers and an A101 card I had laying around. I did a test this weekend and pluged in our PRI in that test server. I never got succeded to have a call trough. When I dialed in, the call is hanged up with : Channel 1/1, span 1 got hanup, cause 6 Spawn extension (ael-default, s, 3) exited non-zero on 'DAHDI/i1/NPANXX-2' Hungup 'DAHDI/i1/NPANXX-2' Here's my dahdi/system.conf : loadzone=us defaultzone=us #Sangoma A101 port 1 [slot:0 bus:6 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 echocanceller=mg2,1-23 hardhdlc=24 my asterisk/chan_dahdi.conf is rather long, but is mostly defaults, with : switchtype=national pridialplan=unknown signalling=pri_cpe group=1 channel = 1-23 as the last non-commented lines. So, for one thing, the card I have in my test server doesn't have an hardware echo canceller, but it's still enabled in my wanpip setting. Could that be a source of problem ? Other than that, is there anything obvious I've missed ? -- _ -- 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] HA Asterisk
On May 1, 2011, at 10:09 AM, Kaushal Shriyan wrote: On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.com wrote: Xorcom makes a box that connects via USB that can do failover. You connect the box to the two system via a USB cable to each system. When the box detects the primary system fails it switches over the the second one. No need for any extra hardware, except a USB cable. http://www.xorcom.com/catalog/xr0015.html http://www.xorcom.com/optional-extras/twinstar.html Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal Correct, there are no cards inside any system. You have an external box that can have a combination of PRI, FXO and FXS ports; depending on need. The external box is connected via USB to the two systems. The twinstar option allows you to connect the external box to two systems via USB and provides fall over from primary to secondary on failure of the primary.-- _ -- 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] Orginate not working well with PSTN lines
Originate successfully queued only means that the originate action was handed off to asterisk not that is was executed yet. There are other events, depending on which events you are reading, that tell you the call was answered and such. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 26, 2011, at 2:43 AM, Ashik Ali wrote: Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890 Response: Success ActionID: ABC45678901234567890 Message: Originate successfully queued -- Remote UNIX connection disconnected Channel DAHDI/1-1 was answered. -- Executing [1234@outbound-ivr:1] SayDigits(DAHDI/1-1, 1234) in new stack -- DAHDI/1-1 Playing 'digits/1.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/2.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/3.gsm' (language 'en') -- DAHDI/1-1 Playing 'digits/4.gsm' (language 'en') -- Executing [1234@outbound-ivr:2] Playback(DAHDI/1-1, demo-congrats) in new stack -- DAHDI/1-1 Playing 'demo-congrats.gsm' (language 'en') -- Executing [1234@outbound-ivr:3] Hangup(DAHDI/1-1, ) in new stack == Spawn extension (outbound-ivr, 1234, 3) exited non-zero on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' Thanks Regards, Ashik -- _ -- 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] DTMF not being sent ( RFC2833 )
I had problems with a system I was trying to bring up using a couple older a104d cards we had lying around. Neither card would pass audio. I worked with one Sangoma tech for a couple hours while he tried various things. The second tech I worked with got on the system and updated the firmware for the cards. When I tried to show him the problem things worked. I said you did something as this did not work an hour ago. He told me the first think he does when troubleshooting is to update the firmware to the current version. A lesson I have now learned. I do that with software but rarely remember to look for firmware updates. Take a look at wiki.sangoma.com and it lets you know current firmware versions as well as how to update if you are not running the current version. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 25, 2011, at 4:41 PM, Edwin Lam wrote: i think i have similar problem after upgraded from 1.4.x to 1.6.2.17. (originally upgraded to 1.8.3.2 unfortunately there were other more pressing problems that forced me to downgraded it to 1.6.2.17) i have a wanpipe device with 2 channels uses PRI signalling to PSTN the other 2 uses FXO signalling (connect to Rhino FXS channel bank). the PRI part works fine but the FXO channels are having DTMF digits skipped. i'm still trying to find out what's wrong with it. On 4/23/11 8:48 AM, David wrote: Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from source. Asterisk : spandsp, dahdi, asterisk. Asterisk-pri : spandsp, libpri, dahdi, asterisk wanpipe I eliminated AGI, hard phones, network et al by setting up this extension : exten = 22,1,Dial(SIP/114186939...@pri1.omnity.net,30,D(132412983 mailto:SIP/114186939...@pri1.omnity.net,30,D(132412983#)) in default. The only other non default setting is in sip.conf I added a outboundproxy ( which does NOT do RTP, only SIP ). I called asterisk from my hard phone ( gxp2000 ) by dialing 22. I see the console DTMF messages indicating the DTMF was sent or received. ( I forgot to keep this output ). I than watch the console DTMF output on asterisk-pri and it showed about half the DTMFs. The pager that was called showed the DTMFs that appeared on the asterisk-pri console. So somewhere between the two machines, the DTMFs have disappeared. So I ran TCPDump on asterisk and saw that close to half of the DTMF events were never sent. tcpdump -i eth0 -n -s 0 dst asterisk-pri-ip -vvv -w ~/dtmf.pcap I imported the file into wireshark on my local machine and confirmed that the dump almost matches what I saw on asterisk-pri. So, problem 1 : Asterisk is not sending all the DTMFs to asterisk-pri. I compared the packet scan to what I saw on asterisk-pri and noticed that between 1 and 3 dtmfs were missing. Problem 2 : Asterisk-pri loses some received DTMFs. I also noticed that some of the DTMFs coming out of asterisk had the wrong Event Duration. I had one DTMF with a duration of about 58000 ( I believe that's 58 seconds ) but I only pressed the button for like 1/3 of a second. What I do not understand is that I in my final test last night was using asterisk 1.6 current with centos ( os that asterisk is developed on from my understanding ) with all default settings ( excluding logger.conf, dialplan and outboundproxy ) and I am having problems with the DTMF. Both servers were installed with CentOS 5.5 and were updated last night, after which I reinstalled asterisk. This did not resolve the issue. I am at wit's end and do not know where to go from here. I would really appreciate it if someone could give me some pointers on where to go next, what additionnal debugging steps I should perform. I would also really appreciate if someone could propose a solution. Please help! David Never give up, never surrender -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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
Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 5:02 AM, Vlasis Hatzistavrou wrote:
Hello,
On an Asterisk 1.4.33.1 in a simple scenario:
[test]
exten = _X.,1,Dial(SIP/12345@peer01,,,)
exten = i,1,Hangup(${HANGUPCAUSE})
exten = t,1,Hangup(${HANGUPCAUSE})
exten = h,1,Hangup(${HANGUPCAUSE})
I have noticed that no matter what value we set in the Hangup(cause code)
commands, if the call is not answered by peer01 for any reason, the actual
cause code returned to the calling party is a 503, no matter what the
${HANGUPCAUSE} is.
Even if we set a fixed value like Hangup(1) (which should give a 404) or
Hangup(17) (which should give a 486), the cause code returned is always a 503.
Has anyone else noticed this? I went through the issue tracker but I couldn't
find any relevant bug posted in the past. I am certain that in previous
versions I could set the reply message to the desired value, so I wonder if
this is a bug in this particular version (1.4.33.1).
--
Best regards,
Vlasis Hatzistavrou.
--
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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)
If what you showed is your whole dialplan then none of the i or t or h
extensions are going to be executed for a non answered call.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 15, 2011, at 6:46 AM, Vlasis Hatzistavrou wrote:
Hello Jim,
Thank you for the reply.
The problem is not reading the ${HANGUPCAUSE} or the ${DIALSTATUS}. It is
that the Hangup(cause) command seems to ignore its argument and just sends
a 503 cause to the caller for all unanswered calls no matter what...
Hangup(cause) was working as expected in previous versions and I wonder if
something was broken along the way that went by unnoticed. I am just asking
in the list in case I am missing something too obvious before posting a bug.
--
Best regards,
Vlasis Hatzistavrou.
On 15/4/2011 4:22 μμ, Jim Dickenson wrote:
My guess is since the call was never answered you should be looking at
${DIALSTATUS}
--
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Re: [asterisk-users] Reach PSTN from another Asterisk
On server B use IAX2/iax-trunk-name/whatever-needs-to-be-dialed-on-A-to-make-call in the Dial command. like Dial(IAX2/sfserver1/9411212) -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 15, 2011, at 12:29 PM, Alejandro Cabrera Obed wrote: Dear, we have the following: - Asterisk A with E1 to PSTN connection. - Asterisk B with IAX trunk to Asterisk A - Outgoing routes between Asterisk A and B - Asterisk A with an outgoing route to PSTN with 9|. dial rule How can I reach the PSTN from Asterisk B through Asterisk A ??? Thanks a lot !!! Alejandro -- _ -- 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] Microsoft Lync server and Asterisk access
We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to connect to an Asterisk box? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card
Do you have the Sangoma wanpipe software installed? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 13, 2011, at 7:37 AM, satish patel wrote: Try dmesg command root@:~# dmesg | grep -i Sangoma [ 2303.473601] WANPIPE(tm) Hardware Support Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.481115] WANPIPE(tm) Interface Support Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.494824] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.506498] WANPIPE(tm) Socket API Module 3.5.19.0 (c) 1994-2010 Sangoma Technologies Inc [ 2303.525334] WANPIPE(tm) WANEC Layer 3.5.19.0 (c) 1995-2006 Sangoma Technologies Inc. From: kaushalshri...@gmail.com Date: Wed, 13 Apr 2011 19:38:06 +0530 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card Hi, I have Sangoma A101DE 1 Port E1/T1 With Hardware Echo Cancellation ( PCI Express ) Card installed on the box. Its not detected. Details are as below :- [root@asterisk ~]# lspci 00:00.0 Host bridge: ATI Technologies Inc RS480 Host Bridge (rev 01) 00:01.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:04.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:05.0 PCI bridge: ATI Technologies Inc RS480 PCI Bridge 00:12.0 IDE interface: ATI Technologies Inc IXP SB400 Serial ATA Controller 00:13.0 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.1 USB Controller: ATI Technologies Inc IXP SB400 USB Host Controller 00:13.2 USB Controller: ATI Technologies Inc IXP SB400 USB2 Host Controller 00:14.0 SMBus: ATI Technologies Inc IXP SB400 SMBus Controller (rev 10) 00:14.1 IDE interface: ATI Technologies Inc IXP SB400 IDE Controller 00:14.3 ISA bridge: ATI Technologies Inc IXP SB400 PCI-ISA Bridge 00:14.4 PCI bridge: ATI Technologies Inc IXP SB400 PCI-PCI Bridge 00:14.5 Multimedia audio controller: ATI Technologies Inc IXP SB400 AC'97 Audio Controller (rev 01) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS480 [Radeon Xpress 200G Series] 01:05.1 Display controller: ATI Technologies Inc Radeon Xpress Series (RS480) 02:00.0 PCI bridge: PLX Technology, Inc. PEX8112 x1 Lane PCI Express-to-PCI Bridge (rev aa) 04:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 20) 05:0a.0 Ethernet controller: Accton Technology Corporation EN-1216 Ethernet Adapter (rev 11) [root@asterisk ~]# cat /etc/redhat-release CentOS release 5.5 (Final) [root@asterisk ~]# asterisk -v Asterisk 1.6.2.11, Copyright (C) 1999 - 2010 Digium, Inc. and others. Created by Mark Spencer marks...@digium.com Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. = Asterisk already running on /var/run/asterisk/asterisk.ctl. Use 'asterisk -r' to connect. [root@asterisk ~]# Please suggest/guide Thanks Kaushal -- _ -- 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] send voicemail to multiple emails
If you want externnotify to not fire when someone checks then put in a new option in voicemail.conf to have it work that way. Then contribute that change and it might be accepted. externnotify_on_check: yes|no or some such thing. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 12, 2011, at 1:52 PM, Steve Edwards wrote: On Tue, 12 Apr 2011, vip killa wrote: Honestly, I don't understand why externnotify should run when someone checks their voicemail... the change i made, makes sense so maybe that should be contributed to the asterisk source. Even if it makes sense to everybody on the list, changes that conflict with documented and implemented behavior that other users may be depending on are unlikely to be accepted. -- 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 -- _ -- 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] Variable inheritance with dialplan command Originate
Another option is to pass the information in the extension. At times I have an
extension like
_[s][o][m][e]-[e][x][a][m][p][l][e].
And call it like some-example:info1:info2 and use cut to extract the info1 and
info2 values. Not real pretty but as this is computer generated calls it gets
the job done.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 8, 2011, at 8:57 AM, Naomi Rosenberg wrote:
Thanks. That's as I thought (feared). Dial is not an option in this case but
I have come up with a workaround involving using a reference number as the
extension and then doing a database call. Not pretty but it works!
Naomi
- Original Message -
From: Sherwood McGowan sherwood.mcgo...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, 8 April, 2011 4:35:43 PM
Subject: Re: [asterisk-users] Variable inheritance with dialplan command
Originate
On 4/8/2011 4:57 AM, Naomi Rosenberg wrote:
Hi,
I would have thought that when spawning a channel using the
Originate() dialplan command, variables prefixed with two underscores
would be preserved.
However this does not work in the following case.
Dialplan code:
[intern]
exten = 200,1,Set(__myvar=foo)
exten = 200,n,Originate(Local/123@test_orig,exten,dummy)
[test_orig]
exten = 123,1,NoOp(${myvar})
exten = 123,n,Hangup()
[dummy]
/end dialplan code.
Console output:
-- Executing [200@intern:1] Set(SIP/200-0018,
__myvar=foo) in new stack
-- Executing [200@intern:2] Originate(SIP/200-0018,
Local/123@test_orig,exten,dummy) in new stack
-- Executing [123@test_orig:1] NoOp(Local/123@test_orig-cbab;2,
) in new stack
-- Executing [123@test_orig:2]
Hangup(Local/123@test_orig-cbab;2, ) in new stack
/end console output.
This is in Asterisk 1.8.3.
Is this expected behaviour or a bug, or am I just confused? I would
appreciate your thoughts on the matter.
Thank you,
Naomi
I believe that it's expected behavior because you're not creating a
child channel, you're originating a different set. Try using Dial
instead of Originate, and you'll get the inheritance behavior you
expected.
-- Sherwood McGowan sherwood.mcgo...@gmail.com
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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Re: [asterisk-users] AMI redirect from Queue to MeetMe
I would be surprised that you did not always hang up the second channel you are
redirecting. Once you transfer one leg there is nothing connected to the second
leg so it goes away, I would think.
What we do is remember the agent number, transfer the caller, and then setup a
call to the agent and meetme room.
More or less like:
Action: Redirect
Channel: SIP/GXP280_18-0001
Exten: do_meetme601MyID
Context: cfmc_cdi_private
Priority: 1
ActionID: MeetMe
Async: true
Action: Originate
Channel: Agent/1001
Exten: do_meetme601MyID2
Context: cfmc_cdi_private
Priority: 1
ActionID: DirectMeet
Async: true
exten = _do_meetme.,1,UserEvent(BeforeMeetMe,Info:${EXTEN:9} ${UNIQUEID}
${CHANNEL})
exten = _do_meetme.,n,Answer()
exten = _do_meetme.,n,Set(CfMC_RoomToUse=${EXTEN:9:3})
exten = _do_meetme.,n,Set(CfMC_CurrentID=${EXTEN:12})
exten = _do_meetme.,n,Set(MEETME_MOH_CLASS=meetme-music)
exten = _do_meetme.,n,MeetMe(${CfMC_RoomToUse},CMpqx1)
exten = _do_meetme.,n,UserEvent(AfterMeetMe,ActionID:${CfMC_CurrentID}
Room:${CfMC_RoomToUse} ${UNIQUEID} ${CHANNEL})
exten = _do_meetme.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 28, 2011, at 1:23 AM, Deka, Rajib IN MAA SL wrote:
Hello List,
I have scenario as follows,
A call comes to queue.
Available agent will answer the call.
BridgeEvent wil be generated in AMI with channel1 and channel2.
Parse channel1 and channel two from the event and redirect them to a meetme
room,
Dialplan,
Exten = 1234,1,MeetMe(1234,1dq)
But sometime it works and sometime one leg gets disconnected after
redirection. Is it a bug to asterisk-1.6.2.13 ?
Regards,
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.d...@siemens.com
Important notice: This e-mail and any attachment there to contains corporate
proprietary information. If you have received it by mistake, please notify us
immediately by reply e-mail and delete this e-mail and its attachments from
your system.
Thank You.
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Re: [asterisk-users] Executing shell commands via AMI
If you want total control from AMI then point at an extension that you can set variables to commands and arguments, call an AGI and set variables that can be passed back to AMI via user events. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 16, 2011, at 3:03 PM, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vinícius Fontes Sent: Wednesday, March 16, 2011 5:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Executing shell commands via AMI But what about if asterisk running with non-privilege user? Still it is not a good idea. Yes I forgot to say that I also run Asterisk as a regular user, which also helps with security. But I really don't see much of a threat on this. AGI does almost the same. This won’t help but I’ll chip in anyway. In AGI, you have “total” local control. In AMI, it’s a crap shoot. -- _ -- 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] How do you handle queues with AMI?
What we do is just before the call to queue we do a userevent that has the uniqueid and the channel and any other information we care about. You can hold on to this information and match it when you get the agentconnect event. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Mar 11, 2011, at 7:21 AM, Danny Nicholas wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Louis Carreiro Sent: Friday, March 11, 2011 9:17 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How do you handle queues with AMI? Hey all, I’m in the process of writing a few applications that are going to either monitor the queue (number of calls, positions, etc) or respond to answering a queue call (if you answer, a window pops up with info about caller, hold time, etc.). I’m writing this in C# but language isn’t important. I’m not looking for a hand out on code, what I’m really interested in is theory or logic. How are other people watching the call come into the queue and watch it from there. What events are you watching? I’ve already got the app to recognize the “packets” of information from the AMI so I can handle them accordingly. I know how to action off of the AgentConnect part but what I’m missing is how to tie that back into the call (Caller ID, etc.). I know the first response will be use the Uniqueid for the call but how? What are your methods for tracking it? How do you know it even entered the queue? Also, as I’m writing this, if anyone would like to help out or share code I’m up for it. I’ll make my code available to all interested in doing this in C# (it’s pretty painless). Thanks! Louis If you look through your CDR, you’ll see the information you need to develop this methodology. Keep in my that (as I understand it), when an agent picks up a call, the uniqueid will change just like the call had been transferred. -- _ -- 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] ChanSpy with alphanumeric SIP channels [1.6.2]
I think in the chanspy application you can give it a template to prepend to
what is entered. If you do chanspy(ab_) you might be able to enter the
remaining digits.
Short of that you can set up a loop that reads the digits, calls
chanspy(ab_${digits}), if the version you are using has my S option then * will
exit the chanspy app and you can loop back to the top.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 9, 2011, at 6:28 PM, Raj Mathur (राज माथुर) wrote:
Hi,
I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5
digits). ChanSpy is working fine for listening in to conversations
initiated by these channels, and I can use '*' to randomly switch
channels. However, is there any way in this scenario to be able to
switch ChanSpy to a specific channel by typing in a ...# key sequence
during a spy session?
Regards,
-- Raj
--
Raj Mathurr...@kandalaya.org http://kandalaya.org/
GPG: 78D4 FC67 367F 40E2 0DD5 0FEF C968 D0EF CC68 D17F
PsyTrance Chill: http://schizoid.in/ || It is the mind that moves
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[asterisk-users] Connecting to Cisco Iad2430 to Asterisk
Is it possible to SIP trunk to this Cisco device so that phones connected to the Cisco box can dial extensions on the Asterisk box? What I would like to be able to do is dial some extension(s) on phones connected to the Cisco box and have the call routed into extension(s) on the Asterisk box. One of our clients has a call center with 65 analog phones connected to the Cisco box. We would like to be able add our dialer appliance into their operation without having to replace any more equipment than needed. We need an easy way for the agents to connect to an extension on our appliance that basically does an agentlogin. Ideas as to how to best accomplish this would be appreciated. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Determine When Call Is Picked Up In Queue
You might be able to use a macro on the dial command (option M) which gets run when the remote end answers. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 29, 2011, at 10:30 AM, Joseph Begumisa wrote: Hi, I have a situation where a call comes in to my asterisk server, goes through an IVR and is then handed off to another asterisk server where it enters a queue waiting for an agent to answer the call. (I do not control the second asterisk server). Is there a way for me to know when the call is actually picked up on the second asterisk server? I have a billing application that needs to start billing when the call is actually answered by an agent. Thanks a lot. Best Regards, 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 -- _ -- 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 check a number online or offline
You can always place a call to an extension that sends a user event from AMI. If there are no native AMI commands that can return what you want originate a call to a local extension that returns a user event. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote: Thanks Dhaval, My purpose is that i want to use java application (using Asterisk Manager Interface) to check a number online, offline or unreachable. Your suggest uses function DEVICE_STATE but this is written in dialplan not application java. Do you know other way to do this for me?thanks and looks forward to listening your reply. Regards! Phuong On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: Hello , You can use Dialplan function DEVICE_STATE, which will gives you perfect status of DEVICE. regards Dhaval On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.net wrote: On 10 Jan 2011, at 10:37, Phuong Hoang wrote: I found the link you have just sent to me but it do`nt help me to resolve this. Can you say clearlier for me? Not really. It's a list of manager commands. There is 'SIPshowpeer' which will work for sip stuff. Try the command 'Command' action and you can send any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work in some cases.. 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 -- _ -- 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] Call Back on Busy
It should not be too hard to write some dialplan code that detects the busy, plays a sound file asking if you want to camp-on to the called device, read an answer and loop around checking device status and placing a call when both the calling device and called device are free. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 10, 2011, at 8:39 AM, John Novack wrote: That function in the telephony world is called camp-on Can't say for sure if Asterisk can do that, not which version, nor freepbx John Novack Ron wrote: Hi All, One of our user asked the question, when she tries to call another local extension but the other end is engaged she will keep on trying until she finally can get thru. So she asked would it be possible to request for an auto-callback from the user she's trying to call to once it's not engaged anymore. is this possible on asterisk? what is that feature called? i am using asterisk 1.4 with freepbx. Thanks in advance. Regards Ron -- _ -- 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 -- Dog is my Co-pilot -- _ -- 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] How to check a number online or offline
If you do an AMI packet like this:
Action: Originate
Channel: Local/get_i...@some_context
Exten: do_noop
Context: some_context
Priority: 1
ActionID: GetInfo
Async: true
and then have a couple extensions that do what you want. Here is what I do in
my case:
exten = get_info,1,Answer()
exten = get_info,n,UserEvent(GetInfo,Version:ABE
DateTime:${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)} CfMC:83351)
exten = get_info,n,Hangup()
exten = do_noop,1,Answer()
exten = do_noop,n,Wait(1)
exten = do_noop,n,Hangup()
You would then do what you need to do in your extensions.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2011, at 6:16 PM, Phuong Hoang wrote:
Thanks Jim,
Can you say about your idea clearlier? I want to use AMI in an application
java to check a number online, offline or unreachable and result is returned
to the appliction java. If the number is online now, i will use AMI to hangup
it, else i do nothing.
Best regards,
Phuong.
On Mon, Jan 10, 2011 at 8:50 AM, Jim Dickenson dicken...@cfmc.com wrote:
You can always place a call to an extension that sends a user event from
AMI. If there are no native AMI commands that can return what you want
originate a call to a local extension that returns a user event.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 10, 2011, at 7:48 AM, Phuong Hoang wrote:
Thanks Dhaval,
My purpose is that i want to use java application (using Asterisk Manager
Interface) to check a number online, offline or unreachable. Your suggest
uses function DEVICE_STATE but this is written in dialplan not application
java. Do you know other way to do this for me?thanks and looks forward to
listening your reply.
Regards!
Phuong
On Mon, Jan 10, 2011 at 3:13 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com
wrote:
Hello ,
You can use Dialplan function DEVICE_STATE, which will gives you perfect
status of DEVICE.
regards
Dhaval
On Mon, Jan 10, 2011 at 4:11 PM, Steve Howes steve-li...@geekinter.net
wrote:
On 10 Jan 2011, at 10:37, Phuong Hoang wrote:
I found the link you have just sent to me but it do`nt help me to resolve
this. Can you say clearlier for me?
Not really. It's a list of manager commands. There is 'SIPshowpeer' which
will work for sip stuff. Try the command 'Command' action and you can send
any CLI command, like sip/iax2 show peers etc. 'ExtensionState' might work
in some cases..
S
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Re: [asterisk-users] Benefit of PRI vs SIP trunk calls
I am running version 1.4.x. Where do I get PRICAUSE? I tried making a call that was not answered and I did not see any more information. The dumpchan of DADHI/23-1 did not happen as that is in a macro that only gets called for an answered call. I only see this: Executing [91112223...@empl:8] Dial(SIP/mine-0521, Dahdi/G1/111222|60|gM(out-dial)) in new stack DEBUG[4907]: dsp.c:1682 ast_dsp_set_busy_pattern: dsp busy pattern set to 0,0 -- Requested transfer capability: 0x00 - SPEECH -- Called G1/111222 DEBUG[3188]: chan_dahdi.c:10135 pri_dchannel: Queuing frame from PRI_EVENT_PROCEEDING on channel 0/23 span 1 -- DAHDI/23-1 is proceeding passing it to SIP/mine-0521 -- DAHDI/23-1 is ringing DEBUG[3188]: chan_dahdi.c:1790 dahdi_enable_ec: Echo cancellation already on -- DAHDI/23-1 answered SIP/mine-0521 -- Executing [...@macro-out-dial:1] DumpChan(DAHDI/23-1, ) in new stack Dumping Info For Channel: DAHDI/23-1: Info: Name= DAHDI/23-1 Type= DAHDI UniqueID= sys.domain.com-1294514614.2630 CallerID= 9111222 CallerIDName= (N/A) DNIDDigits= (N/A) RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x4 (ulaw) WriteFormat=0x4 (ulaw) ReadFormat= 0x4 (ulaw) 1stFileDescriptor= 35 Framesin= 189 Framesout= 176 TimetoHangup= 0 ElapsedTime=0h0m4s Context=macro-out-dial Extension= s Priority= 1 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: MACRO_DEPTH=1 MACRO_PRIORITY=1 MACRO_CONTEXT=from-outside MACRO_EXTEN= DIALEDPEERNUMBER=G1/111222 TRANSFERCAPABILITY=SPEECH DEBUG[4907]: app_macro.c:379 _macro_exec: Executed application: DumpChan DEBUG[4907]: app_dial.c:1927 dial_exec_full: Macro exited with status 0 DEBUG[4907]: chan_dahdi.c:3464 dahdi_setoption: Set option AUDIO MODE, value: ON(1) on DAHDI/23-1 DEBUG[4907]: chan_dahdi.c:3092 dahdi_hangup: Not yet hungup... Calling hangup once with icause, and clearing call DEBUG[4907]: chan_dahdi.c:3460 dahdi_setoption: Set option AUDIO MODE, value: OFF(0) on DAHDI/23-1 -- Hungup 'DAHDI/23-1' == Spawn extension (empl, 9111222, 8) exited non-zero on 'SIP/mine-0521' -- Executing [...@empl:1] Verbose(SIP/mine-0521, 2|Hangup SIP/mine-0521 with cause 16) in new stack == Hangup SIP/mine-0521 with cause 16 -- Executing [...@empl:2] DumpChan(SIP/mine-0521, ) in new stack Dumping Info For Channel: SIP/mine-0521: Info: Name= SIP/mine-0521 Type= SIP UniqueID= sys.domain.com-1294514614.2629 CallerID= 444555 CallerIDName= Jim Dickenson DNIDDigits= 9111222 RDNIS= (N/A) State= Up (6) Rings= 0 NativeFormat= 0x2 (gsm) WriteFormat=0x2 (gsm) ReadFormat= 0x2 (gsm) 1stFileDescriptor= 65 Framesin= 248 Framesout= 253 TimetoHangup= 0 ElapsedTime=0h0m0s Context=empl Extension= h Priority= 2 CallGroup= PickupGroup= Application=DumpChan Data= (Empty) Blocking_in=(Not Blocking) Variables: DIALSTATUS=ANSWER DIALEDTIME=5 ANSWEREDTIME=1 RTPAUDIOQOS=ssrc=671389293;themssrc=651772178;lp=0;rxjitter=0.001217;rxcount=248;txjitter=0.00;txcount=252;rlp=0;rtt=0.00 BRIDGEPEER=DAHDI/23-1 DIALEDPEERNUMBER=G1/111222 DIALEDPEERNAME=DAHDI/23-1 MACRO_DEPTH=0 RCStatus=0 MyChan=SIP sipcallid=0b69233cd5469...@192.168.0.16 SIPUSERAGENT=Grandstream GXP2000 1.2.2.6 SIPDOMAIN=sys.domain.com SIPURI=sip:m...@00.00.000.000:5064 -- Executing [...@empl:3] ExecIf(SIP/mine-0521, 0|Set|DB(conf//haveadmin)=no) in new stack -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 7, 2011, at 12:44 PM, C F wrote: PRICAUSE will give you lots of info on why a call was hungup on. Not sure if SIP will give you the same. On Thu, Jan 6, 2011 at 9:06 AM, Jim Dickenson dicken...@cfmc.com wrote: Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine
[asterisk-users] Benefit of PRI vs SIP trunk calls
Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can access that can know the call was to a fax machine? If a call is placed to a number that is disconnected so a special information tone is played can either a PRI call or a SIP call know this without analyzing the audio stream? Are there reasons to prefer the use of PRI over SIP or SIP over PRI? I would like people's opinions as to if one form is better than the other in any meaningful way. Thanks for you feed-back. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] Moving asterisk from one network to another.
If you set bindaddr in any conf file you will need to change the IP address there. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 24, 2010, at 4:30 AM, Asterisk Man wrote: Friends, Do we need to change any Asterisk configuration files (Or any file related to Asterisk for that matter) when we put Asterisk box from one network to another? It is assumed that DB is on the same box. Asterisk box has got Asterisk running in it with no issues. Probably, it should not complain. I tried to check for IP address in Asterisk files (using ‘find . | xargs grep 192.168.X.XX –sl’), but it seems that Asterisk does not store specific IP in file(s). Your thoughts on this if I m missing something. -AsteriskMan -- _ -- 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] DIALSTATUS on CANCEL
If on the dial command you add option g, if the call is not answered, it will
fall through to the next statement which can be a hangup command and then it
will go to the h extension. If that does not then make the statement after the
dial command a goto h extension.
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On Dec 24, 2010, at 6:03 AM, brya...@zktech.com wrote:
If a call is hung up before an answer our h extension is not running in our
dial macro
Bryant
On Dec 24, 2010, at 3:47 AM, Vardan Harutyunyan hvarda...@gmail.com wrote:
Hello Bryant
Extension h is worked in any case of hangup.
It not important to that the call was answered or no.
It also be more flexible, if you use instead of ${DIALSTATUS}use
${HANGUPCAUSE}, while in most of cause ${DIALSTATUS} is returned the same
return code.
http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
--
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Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
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Bryant Zimmerman wrote:
Vardan
I have not use AEL so it is a bit hard to follow with the formatting the
way it is but it looks like correct.
Please note the h extension only appears to run if a call is connected
so I do not know when the CANCEL would ever be set.
There may be someone else who can speak to this. It also appears thet
${DIALSTATUS} may not be set if the call is not allowed to time out or
dialed. To me it would make sense to set the inital state of the
${DIALSTATUS} to CANCEL and if nothing changes it that would stand, but
I may be missing the point on this can anyone else speak to it?
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Thursday, December 23, 2010 2:11 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
I have make test in AEL.
context fu {
_000./userN = {
Dial(SIP/${EXTEN:3...@prov);
Noop(${DIALSTATUS});
};
h = {
Noop(${DIALSTATUS});
};
};
And look CLI
-- Executing [00018185402...@fu:1] NoOp(SIP/userN-b6317738, )
in new stack
-- Executing [00018185402...@fo:2] Dial(SIP/user3-b6317738,
SIP/18185402...@prov) in new stack
-- Called 18185402...@prov
-- SIP/Prov-082a83b8 is making progress passing it to
SIP/userN-b6317738
== Spawn extension (fu, 00018185402020, 2) exited non-zero on
'SIP/user3-b6317738'
-- Executing [...@fu:1] NoOp(SIP/userN-b6317738, CANCEL) in new stack
I think, I am right
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Bryant Zimmerman wrote:
The Dial Status is not set when accessing it from the h extension.
Bryant
*From*: Vardan Harutyunyan hvarda...@gmail.com
*Sent*: Wednesday, December 22, 2010 10:39 AM
*To*: asterisk-users@lists.digium.com
*Subject*: Re: [asterisk-users] DIALSTATUS on CANCEL
Try to use h extension
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Michael wrote:
Hi Nikhil,
Both debug and verbose are set to 20. That's all I got, but as you can
see, for the other types of reasons, the DIALSTATUS got a value (and we
see the events). I'm pretty sure it's a bug.
Michael
On Wed, Dec 22, 2010 at 9:01 AM, Nikhil d.nik...@cem-solutions..net
mailto:d.nik...@cem-solutions.net wrote:
Hi
Enable debug level to more than 1 ,you may get something.
Thanks
Nikhil
On 12/22/2010 11:26 AM, Michael wrote:
Spawn extension (incoming-private, , 3) exited non-zero
on 'SIP/Proxy-0031'
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Re: [asterisk-users] Asterisk SIP attacks and sshguard
I do not have log examples to provide but do have info about other issues. There is a nocolor option in asterisk.conf that can turn off color. logger.conf has a provision to use syslog directly. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 9, 2010, at 5:57 AM, Joe Greco wrote: Hello, We had been seeing SIP-guessing attacks on our Asterisk server here. While it wasn't that hard to write a once-a-minute cron job to spank the lusers, that runs once a minute and creates little spikes in the usage and I/O graphs, and is slower to respond than I'd really prefer. I felt that it'd be much cooler to get something more comprehensive put together. We don't use fail2ban because I don't like having to install python. sshguard is a high-performance compiled C application that can run off a log file or a pipe from syslogd to sshguard, meaning that it can respond a lot more quickly than once a minute, and works with very modest overhead on the host system. It also has features such as touchiness, so that it can get tougher on a miscreant as time goes on; my own shell script is naive in that once it passes a threshold, there's just a permanent rule generated. This worries me if I ever have a situation where a legitimate remote client gets messed up and tries the wrong password or something like that; sshguard does a much nicer job in this regard. In any case, my initial attempts to create rules for sshguard didn't work right, quite possibly because I don't often work in LEX/YACC. I submitted a request to the sshguard guys suggesting new rules. http://www.sshguard.net/support/submission/detail/49ce7182028d8b6f3e3d/ and on their mailing list, a little more: http://sourceforge.net/mailarchive/forum.php?thread_name=F4E10075-5D93-43B4-B73A-1FD217BE130D%40sshguard.netforum_name=sshguard-users In particular, they're looking for log examples of some of those messages, but I have no idea how to generate the conditions that would cause these messages. I'm also not sure if there's a way to disable color codes in the Asterisk log files; we log indirectly via BSD's logger # asterisk -vvv 21 | logger -t asterisk so it may be thinking that the console is color-capable. We use this method because this forces them through the syslog mechanism; we need that for centralized logging, and it's handy for things like sshguard too. Specifically looking for examples of (or how to generate) 1).*No registration for peer '.*' (from HOST) 2).*Host HOST failed MD5 authentication for '.*' (.*) 3).*Failed to authenticate user .*@HOST.* If anyone who is more familiar with the attacks or how to generate these messages would give me some assistance, or chime in on the sshguard-users list, that'd be most appreciated. Thanks. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. -- _ -- 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] What are the minimal permissions required to read the PeerStatus and Registry events?
A problem that I have always had with AMI events is that they are not controllable at a very fine level. As an example, turning on call class gives WAY more that one might always want. I had posted a patch some time ago that added a new class. The patch was rejected with a comment that some work would be undertaken for 1.8 that would allow finer control. As I am using the ABE version of asterisk I have not looked at 1.8 to see if anything has been done. It would be very useful if there was more control over which events one sees. As I recall all the events have a name and if one could somehow say I want to see events with names x,y,z and forget the whole class mess. That way if you want to see some specific events but not others you could get what you want. I know this does not answer you question but I for sure feel your pain. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Dec 3, 2010, at 10:36 AM, Frank Church wrote: I am logging events from the AMI and the PeerStatus and Registry events show that the privilege for them is System,All. Can a lower set of privileges be used? All looks pretty high to me. /Frank -- _ -- 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] How to hangup all channels
Can also do asterisk -r -x 'restart now'
asterisk*CLI help restart
restart gracefully Restart Asterisk gracefully
restart now Restart Asterisk immediately
restart when convenient Restart Asterisk at empty call volume
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On Nov 27, 2010, at 8:45 AM, Steve Edwards wrote:
On Sat, 27 Nov 2010, Giuseppe D'alessio wrote:
Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels.
1) sudo /etc/init.d/asterisk restart
2) Write a script to do asterisk -r -x 'core show channels', parse the
output and do asterisk -r -x 'channel request hangup ${CHANNEL_NAME} for
each channel.
3) Write a script to do #2 using AMI.
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Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Installing Asterisk to it's own directory
What you did is what I would have done. That way the executables have their conf file location adjusted and everything will be inside the specified --prefix location. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 20, 2010, at 5:48 AM, Stephen Brown wrote: Thanks... I actually did a ./configure --prefix=/root/asterisk18 and ended up with this: -- _ -- 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] DAHDI vs mISDN was Re: Asterisk, VoIP and Samsung iDCS100
For sure DAHDI and libpri support is there for Asterisk 1.4.x. I am not sure either are tied to a specific version of Asterisk as it is the chan_dahdi module that interfaces with DADHI. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 14, 2010, at 8:38 AM, Gordon Henderson wrote: On Wed, 3 Nov 2010, Gordon Henderson wrote: On Wed, 3 Nov 2010, Philipp von Klitzing wrote: Hi! Side note: Stay away from solutions that use mISDN, instead go with Zaptel (DAHDI), Woomera or CAPI. Interesting. I've been usng mISDN for some years now without issues. Why should I migrate to DAHDI? None - if you are happy then don't touch it. :-) Otherwise search this list's archive. Ah, that would be too easy ;-) However I am in the process of upgrading an number of systems from 1.2+mISDN to 1.4 + ... So maybe I'll go and have a look at using DAHDI since I think I've almost got the hand of it for analogue and PRI systems now... So I'll go and do some looking - but you reckon DAHDI and ISDN2e (UK: BRI) is as stable/usable as mISDN might be? Well, just to follow this up - it looks like there is no DAHDI and BRI support in asterisk 1.4 at all. libpri has support in 1.6, but not 1.4, so it's mISDN for the time being. (unless someones knows otherwise) Gordon -- _ -- 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] dial plan and sip
You get into asterisk by saying asterisk -r. You then up the verbosity by saying core set verbose 3 or some such number. You the call your number and you should see the steps of your dialplan execute. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Nov 13, 2010, at 7:02 PM, Thomas Perron wrote: How do I see the error message? the phone call seemed to get through but I did not see anything on my 1.4 console. i used 1.6.x before. having trouble with this for some reason. older stuff. i have one session open at the prompt but nothing shows up. On Sat, Nov 13, 2010 at 9:53 PM, Brett Woollum br...@woollum.com wrote: What is the error message? Sent from my iPhone On Nov 13, 2010, at 6:28 PM, Thomas Perron thomas.per...@gmail.com wrote: Hi Brett, It did not work. I will try other ideas. SIP or Dial plan problem? registeration? On Sat, Nov 13, 2010 at 8:55 PM, Brett Woollum br...@woollum.com wrote: Try changing this line: exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) To: exten = s,n,Dial(SIP/1703...@jazzey,120,A,(demo-thanks)) Sent from my iPhone On Nov 13, 2010, at 5:38 PM, Thomas Perron thomas.per...@gmail.com wrote: Here is a very very basic config. But, not working (: I simply want to dial the DID that is registered with the SIP provider. then, as you can see the call should dial the 703111 number Hints please? sip.conf ;register = 908366554:396...@carrier.jazzey.com register = 908366554:396...@sip.jazzey.com [jazzey] type=friend host=sip.jazzey.com username=908366554 secret=396444 qualify=no insecure=invite extensions.conf exten = s,1,Answer() exten = s,n,Wait(2) exten = s,n,Dial(SIP/jazzey/1703111,120,A,(demo-thanks)) exten = s,n,Wait(2) exten = s,n,Hangup() -- _ -- 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 -- _ -- 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] Get the Uniqueid of Action Originate in the AMI
The other thing you can do is put UserEvent() calls in your dialplan that can
have pretty much anything you want in them.
exten = s,5,UserEvent(DidQueue,${UNIQUEID} ${CHANNEL})
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Nov 8, 2010, at 10:45 AM, Miguel Molina wrote:
El 08/11/10 13:12, Rodrigo Lang escribió:
Hi to all.
I'm begin a use the AMI and i have the need to get the uniqueid from the
call i have generate using the Action Originate. Anyone can help me?
When I generate these commands:
action: Originate
channel: SIP/101
application: Dial
data: SIP/100,120,Ttr
The only response I get when the call is answered, is this:
Response: Success
Message: Originate successfully queued
Hi,
If you are using the originate action in asynchronous mode, you will receive
the uniqueid of the originated call in the OriginateResponse event, not in
the response of the action.
Regards,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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Re: [asterisk-users] DAHDI FXO port only recognizes the S extension
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 29, 2010, at 10:20 AM, A J Stiles wrote: On Wednesday 29 Sep 2010, Songtao Yu wrote: Hi All, When I tried to write my dial plan as below for my FXO port, which connects one PSTN line: [from-pstn] exten =s,1,Answer() exten =s,n,Wait(1) exten =_X.,1,Dial(DAHDI/1) exten =_X.,n,Hangup I got the following message: Connected to Asterisk 1.6.2.13 currently running on fax (pid = 8154) Verbosity was 0 and is now 4 -- Starting simple switch on 'DAHDI/1-1' -- Executing [...@from-pstn:1] Answer(DAHDI/1-1, ) in new stack -- Executing [...@from-pstn:2] Wait(DAHDI/1-1, 1) in new stack -- Auto fallthrough, channel 'DAHDI/1-1' status is 'UNKNOWN' -- Hungup 'DAHDI/1-1' But if I changed the _X. to S extension, I can get the whole thing to work well: [from-pstn] exten =s,1,Answer() exten =s,n,Wait(1) exten =s,n,Dial(DAHDI/3) exten =s,n,Hangup Would you please let me which casuses this issue? Extensions represent different numbers dialled by the calling party. An FXO port has only *one* number associated with it -- the number of the POTS line to which it is connected. It does not, therefore, have to be able to differentiate between extensions. Incoming calls just go straight to the s extension of the context associated with the channel. If for some reason you have more than one FXO port (ordinarily, you would get multiple lines by means of ISDN), then just bring each one in on a separate context. Either that or look at the channel in the dialplan and do what you want based on which channel the call was received on. -- AJS -- _ -- 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] Problems compiling Asterisk on Debian
Did you install the header files after ./configure was run? If so redo the ./configure command and see what that does. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: Hello Paul, Here is the output of the commands: r...@sangoma-testing:/home# ls -la /lib/modules/ total 12 drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 r...@sangoma-testing:/home# ls -la /usr/src/linux lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux - linux-headers-2.6.26-2-amd64 Seems to be OK, isn't? Thanks! 2010/9/27 Paul Belanger paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias ing.diasda...@gmail.com wrote: The same problem! What is the output from the following? $ ls -la /lib/modules/ $ ls -la /usr/src/linux -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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 -- _ -- 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] Problems compiling Asterisk on Debian
Do you not need to do a ./configure command before make make install? If so issue the ./configure command again and see if that fixes the problem. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 27, 2010, at 4:32 PM, Danny Dias wrote: Thanks Jim, What do you mean with redo ? I did not run the ./configure, i'm installing dahdi-linux and just need : make make install The problem is when i issue make Thanks for your answer my friend! 2010/9/28 Jim Dickenson dicken...@cfmc.com Did you install the header files after ./configure was run? If so redo the ./configure command and see what that does. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 27, 2010, at 3:57 PM, Danny Dias wrote: Hello Paul, Here is the output of the commands: r...@sangoma-testing:/home# ls -la /lib/modules/ total 12 drwxr-xr-x 3 root root 4096 2010-09-24 10:21 . drwxr-xr-x 13 root root 4096 2010-09-27 12:57 .. drwxr-xr-x 4 root root 4096 2010-09-27 09:06 2.6.26-2-amd64 r...@sangoma-testing:/home# ls -la /usr/src/linux lrwxrwxrwx 1 root src 28 2010-09-27 12:26 /usr/src/linux - linux-headers-2.6.26-2-amd64 Seems to be OK, isn't? Thanks! 2010/9/27 Paul Belanger paul.belan...@polybeacon.com On Mon, Sep 27, 2010 at 1:09 PM, Danny Dias ing.diasda...@gmail.com wrote: The same problem! What is the output from the following? $ ls -la /lib/modules/ $ ls -la /usr/src/linux -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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 -- _ -- 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 -- _ -- 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] Playing Audio To One Channel
One way to do it is to use ChanSpy and the whisper option. We use AMI to play
sound bits to one leg of the call.
Something like
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 20, 2010, at 5:51 AM, Jon Farmer wrote:
Hi
I have a call established and I want to play audio to just one channel
on that call. Is this possible? If so, how? My google-fu has failed on
this one.
Regards
Jon
--
Jon Farmer
Tel 07795 118140
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Re: [asterisk-users] ChanSpy getting piled up
chanspy as best I can tell from the code will not lock on a single device and
when that device goes away exit. What is passed to chanspy is a template for a
channel name. I submitted a patch to add option s so that chanspy would stop
when the one channel I wanted to watch went away or I used * to stop.
https://issues.asterisk.org/view.php?id=14594
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Sep 1, 2010, at 6:24 AM, Rushikesh wrote:
Hi list,
Im using asterisk 1.6.0.10 and have following dialplan for doing chanspy
[app-chanspy]
include = app-chanspy-custom
exten = 555,1,Read(SPYNUM,extension)
exten = 555,2,ChanSpy(SIP/${SPYNUM},q)
exten = 555,n,Hangup
but if the channel is hang up or even destroyed the chanspy is not
getting killed.
asteriskcore show channels verbose
.
.
.
SIP/1009-b6c5b398from-internal555 2 Up
ChanSpy SIP/1002,q1009
554:53:1 (None)
SIP/1009-b5004908from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-19 (None)
SIP/1009-b50a4e30from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-9: (None)
SIP/1009-b50702a8from-internal555 2 Up
ChanSpy SIP/1002,q1009
-571:-5: (None)
SIP/1009-09bafcd0from-internal555 2 Up
ChanSpy SIP/1002,q1009
-570:-57 (None)
.
.
.
Is there a way to cleanup this ?
Regards
Rushikesh
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Re: [asterisk-users] ChanSpy getting piled up
I had the same need which is why I submitted the patch. I think the feature might finally be added to 1.8, it I remember correctly. I am not aware of any other way around this. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Sep 1, 2010, at 9:12 AM, Rushikesh wrote: On Wednesday 01 September 2010 09:01 PM, Jim Dickenson wrote: chanspy as best I can tell from the code will not lock on a single device and when that device goes away exit. What is passed to chanspy is a template for a channel name. I submitted a patch to add option s so that chanspy would stop when the one channel I wanted to watch went away or I used * to stop. https://issues.asterisk.org/view.php?id=14594 Hi Jim, Thanks for your reply, Im a new user to asterisk and have very basic knowledge of it. by looking at patch I think you are suggesting me to apply the patch to asterisk source code and recompile my asterisk. Actually this is a production system so I'm not sure whether my Boss will allow me to do it ;) . Do you know any other work around for this ? As you said I need to stop chanspy once the channel wen away. Regards, Rushikesh -- _ -- 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] Playback during call
Your ami packet is not setting the w option for chanspy, nor I am sure you can
do this.
You might want to create an additional exten that takes a variable from your
ami packet and does the chanspy that way.
I use an ami packet like this with extension that do the work.
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Aug 9, 2010, at 5:19 PM, Gabriel Ortiz Lour wrote:
Hi all,
How can I playback a file within an active call?
I've tried with ChanSpy whisper mode like this (using AMI):
Action: Originate
Channel: Local/9...@default
Priority: 0
Variable: MSG=test
Application: ChanSpy
Data: SIP/1234-123
Async: 1
and in the dialplan:
[default]
exten = ,1,Answer()
exten = ,n,Wait(2)
exten = ,n,Playback(${MSG})
Where SIP/1234-123 is the up bridged channel.
But this is not working (it seams that will work on the rolling CLI, but no
sound at all)
Is there a better way to do it?
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Re: [asterisk-users] How to track a call result originated from originate AMI command
We track status of calls and many other actions using user events in our dialplan. The dial and queue commands allow for either agi or macros to be executed just before a connection is made. Use option g in dial to allow one to execute a user event after the dial command finishes. Use the h extension to track hang ups. We set an action token variable to a unique value for each originate and all the user events have this token so we can tie them back to the original originate. We turn off most AMI read message classes to cut down on packet volume. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Aug 8, 2010, at 7:18 AM, Nasir Iqbal wrote: Hi, Confusing! you are not alone here. Actually there is no unified development approach exist in Asterisk, every module, application introduce a new way to handle same things!! And the monitoring is most difficult part! you have to write different parsing algos to get each bit of information, and unfortunately you have to rewrite most of your code for every new release! And regarding your question, I recommend you to use AGI for monitoring here is some tips for you in originate command use extension as destination. create failed extension in same context. you can include some variables in originate command which can be used later in dialplan. use AGI scripts in destination and failed extensions to get and save call status in database. Regards On Sun, Aug 8, 2010 at 6:10 PM, thiyagu venkatesan thiyagu.v...@gmail.com wrote: Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were very confusing and I m getting various events with different uniqueid value. For a single call I m getting events with four or five uniqueid. I also filtered using specific channel but also I m getting events with different uniqueid. How can I find the below status for the call generated using originate command through AMI events, 1. Answer 2. No Answer 3. Busy Can any one help me for this. Thanks, Thiyagu VOIP -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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 -- _ -- 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] Urgent help = RUBY AGI
Do you just have one agi you are running? If so that will not work. Your one
agi is hung on the dial step until it finishes at which time the agi will go
away, I think. You need the one agi to cause the dial to occur and another one
to capture the information when the macro runs. You can try adding option g to
the dial command and then retrieve the variable you set in the macro after the
dial step finishes. I do not use agi's much so I might be off base with all
this but it is something to look in to.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 29, 2010, at 7:22 AM, Zarko Zivanovic wrote:
That looks easy. I must say that I am very frustrated as this has took my all
week, and beside dumpling that data via macro I wasnt able to
use that data in the ruby script that we have. I didnt write the script is
something old that we use but i was sure we could add few things in that very
script and continue
to use it. I posted almost all script that we use and it surprised me that no
one was able to find the solution so far.
Zarko
On Thu, Jul 29, 2010 at 3:46 PM, Danny Nicholas da...@debsinc.com wrote:
I can’t even spell RUBY, so I don’t have a clue as to how the AGI works. I
do know a little bit about AGI in general. The way I typically run my AGI’s
is something like this:
exten = 933,1,Answer
exten = 933,n,Set(ABA=02107)
exten = 933,n,Set(city=Birmingham)
exten = 933,n,Set(state=AL)
exten = 933,n,Set(zip=35244)
exten =
933,n,AGI(cityweather.agi,${ABA},${city},${state},${zip},${CHANNEL(language)})
exten = 933,n,hangup()
I’m a PERL weenie, so I can “shell check” my agi’s by going to
/var/lib/asterisk/agi-bin and doing
./cityweather.agi 02107 Birmingham AL 35244 en
And getting back a STDOUT output that simulates what I should get from the
CLI output.
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Re: [asterisk-users] How to record and playback at the same time
Depending on what you are recording there might be two files, one for each leg of a call, until the call ends and the files are mixed. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 29, 2010, at 7:46 AM, Motiejus Jakštys wrote: On Thu, Jul 29, 2010 at 2:32 PM, Benny Amorsen benny+use...@amorsen.dk wrote: Sherwood McGowan sherwood.mcgo...@gmail.com writes: I'm going to go ahead and say that while I'm not one of the developers, I think it's safe to say that you cannot record to a file and play it back at the same time. Probably something like file locking (for the record, locks it from access by other processes, etc)... There is nothing in Unix/Linux which prevents the playback of a file while it is being recorded. File locks in Linux are purely advisory; it is up to the applications whether they choose to respect them. The only challenge is whether the header has been written correctly, and you should be able to do without that in a pinch. What happens if you copy the half-written file to a computer with speakers and try to play it through the speakers? Even (incorrect) headers are not a problem if you know the exact file format. apt-get/yum/pacman/emerge/whatever install/-S/whatever sox man (sox|play) Quick grep through sox source did not yield any fcntl functions, IMHO sox should ignore any locked files and you should be able to play them without problems. Motiejus -- _ -- 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] Nat issue one way audio on IP dial
Do you have your softphone setup to use a stun server so it can send it's public IP address in the SIP packets? I see in the SIP debug output a 192.168 address for the RTP packets to go to which of course will not work. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 28, 2010, at 9:23 AM, Nasir Javaid wrote: hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can hear the called user but the called user can not here the caller voice. If the caller calls the other user by username instead of IP:Port , the voice is perfect both ways. what i have noticed is that IP:Port dial is missing a parameter rinstance in Contact , To headers for adf. what is rinstance for? Also something with Contact header seems fishy. or RTP issue. that is Dial(SIP/adf,30,r) works fine with bothway audio, but Dial(SIP/116.18.35.235:28614,30,r) has one way audio. / \ | | this is IP:Port of of adf please help as it's almost 2 weeks and i have found to suitable answer from any forum. I nead to know what can i do to modify Headers or settings in conf files to correct this problem. Below is the conf of calling user [pepsi] username=pepsi type=friend secret=123456 qualify=yes nat=no insecure=port,invite incominglimit=1 outgoinglimit=1 host=dynamic dtmfmode=rfc2833 context=out canreinvite=yes callerid=pepsi coke 12345678901 accountcode=6:0:pepsi amaflags=default disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm Below is the conf of called user [adf] username=adf type=friend secret=123456 qualify=yes nat=yes insecure=port,invite incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=user canreinvite=yes callerid=adf xyz 11223344556 accountcode=1:0:adf amaflags=default disallow=all allow=g729 allow=ulaw allow=alaw allow=gsm below is my sip debug after dialing Audio is at 79.80.x.x port 16238 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 116.18.35.235:28614: INVITE sip:a...@116.18.35.235:28614 SIP/2.0 Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport From: pepsi coke sip:12345678...@79.80.x.x:5678;tag=as42ec768c To: sip:a...@116.18.35.235:28614 Contact: sip:12345678...@79.80.x.x:5678 Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 21 Jul 2010 15:10:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 285 v=0 o=root 9626 9626 IN IP4 79.80.x.x s=session c=IN IP4 79.80.x.x t=0 0 m=audio 16238 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- [Jul 21 11:10:22] WARNING[23814]: chan_sip.c:2872 sip_call: Setting auto-congest time to 15000 ms. -- Called a...@116.18.35.235:28614 ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: sip:a...@116.18.35.235:28614 To: sip:a...@116.18.35.235:28614;tag=d54e632c From: pepsi cokesip:12345678...@79.80.x.x:5678;tag=as42ec768c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 0 - --- (9 headers 0 lines) --- -- SIP/116.18.35.235:28614-007f4660 is ringing ast-server*CLI --- SIP read from 116.18.35.235:28614 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 79.80.x.x:5678;branch=z9hG4bK46e569df;rport=5678 Contact: sip:a...@116.18.35.235:28614 To: sip:a...@116.18.35.235:28614;tag=d54e632c From: pepsi cokesip:12345678...@79.80.x.x:5678;tag=as42ec768c Call-ID: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 185 v=0 o=- 6 2 IN IP4 192.168.0.12 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.0.12 t=0 0 m=audio 55246 RTP/AVP 8 0 101 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (11 headers 9 lines) --- Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP
Re: [asterisk-users] URgent - capturing 'answered'
Which version of Asterisk are you running?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 2:19 AM, Zarko Zivanovic wrote:
Great, but how exactly do i find that channel - that is my question - which
command.
I am using ruby instead of agi - and i am looking for a command to capture
it in ruby.
I tried this:
# Create a new file and write to it
File.open('log.txt', 'w') do |f2|
# use \n for two lines of text
f2.puts Created by Satish\nThank God!\n my variables are '$loc',
'$agi.get_variable(EXTEN)', '$variable1', '$variable2'
end
$my.query(UPDATE call_log SET endtime = NOW() WHERE
id = #{call_log_id})
- query gets executed, but log.txt wasnt created.
Not to mention that I still didnt manage to catch who answered the call.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Monday, July 26, 2010 7:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
If all you need to do is the the channel name of the channel that answered
the phone why are you doing so much work? Version 1.4 allows for an agi to
be called when the dial command is answered. Version 1.6+ allows an agi as
well as a macro to be called. You can find the channel that answered a multi
channel dial command. Is this not what you wanted to know?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 10:40 AM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] Urgent help = RUBY AGI
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give you
an exact solution but I can tell you that the script that you are using will
not work. In the dial command you need to add the M option which will call a
macro when the call is connected. In that macro you can then find the channel
that answered the call and do what you want from there. You can call another
AGI or set variables or whatever. If agi.exec works like a dialplan step then
the dial step will hang if the call is answered and the agi.get_variable
statement will not execute unless the call was not answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will see on
the CLI all the variables that are available to you. ${CHANNEL} will have SIP/
voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here’s something that should be easy for RUBY pro’s.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it’s executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
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Re: [asterisk-users] Urgent help = RUBY AGI
You can put multiple options in the dial command if that is what you are asking.
And by the way several emails, including a previous one of mine, told you to
use the M option and a macro.
In this email I gave you more detailed information but if you had done core
show application dial on CLI you should have been able to ask more directed
questions.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 9:28 AM, Zarko Zivanovic wrote:
Jim thanks.
I will test this first thing in the morning as I am out of the office now. As
a matter of fact I cant wait to test this, as it has been the first
reasonable thing that looks like it could work.
In the meantime , do you happen to know if there is a way to call both macro
(M) and music on hold (m) in that $agi.exec line?
or is the right thing to do to place moh command in macro?
As I said, I cant wait to try it first thing in the morning and tell you (and
others) how it went. I am sure this will be the good reference to other
people looking for the same thing online as I have found quite a bunch of
similar open threads.
Zarko
On Tue, Jul 27, 2010 at 5:31 PM, Jim Dickenson dicken...@cfmc.com wrote:
I have never used 1.2.9.1 or anything in the 1.2.x range so I can not give
you an exact solution but I can tell you that the script that you are using
will not work. In the dial command you need to add the M option which will
call a macro when the call is connected. In that macro you can then find the
channel that answered the call and do what you want from there. You can call
another AGI or set variables or whatever. If agi.exec works like a dialplan
step then the dial step will hang if the call is answered and the
agi.get_variable statement will not execute unless the call was not answered.
Try
r = $agi.exec('DIAL', SIP/voipuserZap/32Zap/33Zap/34Zap/35,,M(testing))
And then have something like this in extensions.conf
[macro-testing]
exten = s,1,DumpChan()
You will see that this macro runs when the call is answered and you will see
on the CLI all the variables that are available to you. ${CHANNEL} will have
SIP/ voipuser-e989 in your example below.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 27, 2010, at 7:21 AM, Zarko Zivanovic wrote:
Here’s something that should be easy for RUBY pro’s.
Here is a script:
1.times do
r = $agi.exec('DIAL',
SIP/voipuserZap/32Zap/33Zap/34Zap/35)
r = $agi.get_variable('DIALSTATUS')
# $agi.set_variable(' WHOANSWERED ',...)
retry if r.message.include?('BUSY')
end
when it’s executed it shows this in the console:
AGI Rx ANSWER
AGI Tx 200 result=0
AGI Rx EXEC DIAL SIP/voipuserZap/32Zap/33Zap/34Zap/35
-- AGI Script Executing Application: (DIAL) Options:
(SIP/voipuserZap/32Zap/33Zap/34Zap/35)
-- Called voipuser
-- Called 32
-- Called 33
-- Called 34
-- Called 35
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/voipuser-e989 is ringing
-- SIP/ voipuser-e989 answered Zap/1-1
What we need is to be able to populate the variable WHOANSWERED with info
SIP/ voipuser
In this case, or whoever answers next time.
Thanks in advance!
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Re: [asterisk-users] URgent - capturing 'answered'
Depending on the version of Asterisk you are running you can call a macro or an
agi as option to dial. These will be called when the line is answered and you
can find the channel name of who answered.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 5:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime = NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what is
wrong - if you have any idea please help.
Zarko
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Davies
Sent: Monday, July 26, 2010 1:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 3:41 PM, Zarko Zivanovic wrote:
Hello everyone.
I need a quick help on how to capture who answered the call with agi.
Here is an example:
-- Zap/32-1 is ringing
-- Zap/33-1 is ringing
-- Zap/34-1 is ringing
-- Zap/35-1 is ringing
-- SIP/operator1-e77f answered Zap/23-1
So how can I capture this value and write it to mysql?
If you use cdr_mysql, then this data should already be written to the
dstchannel column in the cdr table.
I already have this:
$my.query(UPDATE call_log
SET endtime = NOW() WHERE id = #{call_log_id})
And i needed to do something like:
$my.query(UPDATE call_log
SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id =
#{call_log_id})
Alternatively you may be able to access ${CDR(dstchannel)}.
I've not checked any of the above, but I believe it is right.
Regards,
Steve
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Re: [asterisk-users] 'dirty' upgrade of 1.4
You should be able to compile the new version, stop asterisk then make install. If you do not do make samples then your conf files will be left alone. Once you have done make install you can the start asterisk again. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 26, 2010, at 5:11 AM, Andrew Thomas wrote: Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this routine - or can I just 'upgrade' and then do a * 'restart'? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] URgent - capturing 'answered'
If all you need to do is the the channel name of the channel that answered the
phone why are you doing so much work? Version 1.4 allows for an agi to be
called when the dial command is answered. Version 1.6+ allows an agi as well as
a macro to be called. You can find the channel that answered a multi channel
dial command. Is this not what you wanted to know?
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jul 26, 2010, at 10:40 AM, Zarko Zivanovic wrote:
Hi Andres,
I did try what you said, but it didnt create any files:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2'
/tmp/variables.txt;
system($message);
permissions seem to be fine, echo is in place.
I posted the whole script that i am using in the main thread - if you can
please loook at it.
Zarko.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres
Sent: Monday, July 26, 2010 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 7/26/2010 12:27 PM, Zarko Zivanovic wrote:
I tried this:
loc = $agi.get_variable('EXTEN')
$my.query(UPDATE call_log SET local = #{loc}, endtime = NOW() WHERE id =
#{call_log_id})
When I troubleshoot AGI scripts, I output stuff to text files for
debugging purposes. I suggest you output all your variables to a file
and then you will learn if the variables do have the info you need.
Something like:
$message=/bin/echo my variables are '$loc', '$variable1', '$variable2',
etc /tmp/variables.txt;
system($message);
Andres
http://www.neuroredes.com
No success. Anybody please help!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Monday, July 26, 2010 3:44 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] URgent - capturing 'answered'
On 10-07-26 08:10 AM, Zarko Zivanovic wrote:
Hello Steve and thanks for your answer,
However I tried:
$my.query(UPDATE call_log SET local='#{CDR(dstchannel)}', endtime =
NOW()
WHERE id = #{call_log_id})
And it does write nothing to the database.
I guess there is a error in ruby expression above but I am not sure what
is
wrong - if you have any idea please help.
If that is your literal quote, then I think you need to change the # to a
$
as
Asterisk dialplan functions and variables start with ${ vs #{
Unless that is some special indication in SQL that I'm unfamiliar with.
Leif Madsen.
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Re: [asterisk-users] How to Dialogic 240/JCT-T1 interface with Asterisk?
What do you mean now that ABE is discontinued? My company payed thousands of dollars this year for the product and the support it provides! -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jul 5, 2010, at 5:42 PM, Carlos Chavez wrote: On Mon, 2010-07-05 at 19:59 -0400, Paul Belanger wrote: On Mon, Jul 5, 2010 at 7:29 PM, AMARDEEP SINGH yahhod...@gmail.com wrote: Please share experience if anyone have successfully configured Dialogic JCT-T1 card with asterisk? Your not going to find much; there is no channel driver for Dialogic. Dialogic drivers were only supported in Asterisk Business Edition (ABE) and never in the free version because of proprietary drivers. Now that ABE is discontinued there is no support for Dialogic cards. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] queue command in asterisk 1.4 with macro-argument
Yes it gets called when the call is connected to a queue member. In version 1.4.x you can execute an AGI instead of a sub or macro. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 30, 2010, at 7:20 AM, Jonas Kellens wrote: Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? 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] queue command in asterisk 1.4 with macro-argument
Here is a simple AGI using cagi that creates a user event when a call is
connected with a queue member:
#include stdio.h
#include stdarg.h
#include cagi.h
int main (int argc, char *argv[]) {
AGI_TOOLS agi;
AGI_CMD_RESULT res;
intrtn;
char channel_name[200], uniqueid[200], Interface[200], Event[1000];
rtn = AGITool_Init(agi);
// rtn = AGITool_verbose(agi, res, AGITool_ListGetVal(agi.agi_vars,
// agi_request), 0);
// sprintf(Event, Do verbose= %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
rtn = AGITool_get_variable2(agi, res, CHANNEL,
channel_name, sizeof(channel_name));
// sprintf(Event, Get CHANNEL = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
rtn = AGITool_get_variable2(agi, res, UNIQUEID,
uniqueid, sizeof(uniqueid));
// sprintf(Event, Get UNIQUEID = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
rtn = AGITool_get_variable2(agi, res, MEMBERINTERFACE,
Interface, sizeof(Interface));
// sprintf(Event, Get MEMBERINTERFACE = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
sprintf(Event, DidQueue|\%s %s %s, uniqueid, channel_name,
Interface);
rtn = AGITool_exec(agi, res, UserEvent, Event);
// sprintf(Event, Do UserEvent = %d, rtn);
// AGITool_verbose(agi, res, Event, 0);
AGITool_Destroy(agi);
return 0;
} /* main */
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jun 30, 2010, at 8:31 AM, Jonas Kellens wrote:
Taking my first steps into AGI then :
[r...@asterisk agi-bin]# cat sample.agi
#!/usr/bin/php -q
?php
$MYSQLSERVER2=localhost;
$MYSQLUSER2=user;
$MYSQLPASSWD2=passwd;
set_time_limit(30);
require('phpagi/phpagi.php');
$agi = new AGI();
$db=mysql_connect($MYSQLSERVER2, $MYSQLUSER2, $MYSQLPASSWD2);
mysql_select_db(Asterisk, $db);
$QUERY=SELECT vmcontext FROM AstDB WHERE ID='40';
$agi-verbose(query is: $QUERY, 3);
$result=mysql_query($QUERY);
$VMCONTEXT=mysql_fetch_array($result);
$agi-verbose(VMCONTEXT is: $VMCONTEXT, 3);
$vmcontext=$VMCONTEXT['vmcontext'];
$exten = $agi-request['agi_extension']; //Dialed extension
// the result is stored in $exten
$agi-verbose(variable exten : $exten, 3);
$agi-verbose(variable vmcontext : $vmcontext, 3);
//
?
[Jun 30 17:26:04] -- Executing [...@test:3] AGI(SIP/test-0054,
sample.agi) in new stack
[Jun 30 17:26:04] -- Launched AGI Script
/var/lib/asterisk/agi-bin/sample.agi
[Jun 30 17:26:04] -- sample.agi: query is: SELECT vmcontext FROM AstDB
WHERE klantID='40'
[Jun 30 17:26:04] -- sample.agi: VMCONTEXT is:
[Jun 30 17:26:04] -- sample.agi: variable exten : 123
[Jun 30 17:26:04] -- sample.agi: variable vmcontext :
[Jun 30 17:26:04] -- AGI Script sample.agi completed, returning 0
Does AGI not interpret my query correctly ? As there is no output for
$vmcontext...
Jonas.
On 06/30/2010 04:54 PM, Jim Dickenson wrote:
Yes it gets called when the call is connected to a queue member.
In version 1.4.x you can execute an AGI instead of a sub or macro.
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Re: [asterisk-users] Want to retrieve the value of contact header
You might take a look at the SIPHEADER function which can return specific SIP headers. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 30, 2010, at 7:36 PM, kamrun nahar bina wrote: Dear all, I want to retrieve the value from Contact header and from From header which is 0345001280 from the following two lines: Contact: sip:0345001...@123.50.217.143 From: 99 sip:0345001...@113.34.235.106;tag=as191896a1 Is it possible in asterisk to retrieve that value? I am getting following value in the corresponding variable when I pass the following SIP message. Is there anything which contain the value of 0345001280 of contact ? Corresponding value: CALLERID(num): 185475 CALLERID(name) : 99 SCI-PEERNAME : 185475 SIP message: INVITE sip:08058913...@113.34.235.106 SIP/2.0 Via: SIP/2.0/UDP 123.50.217.143:5060;branch=z9hG4bK100b063a;rport From: 99 sip:0345001...@113.34.235.106;tag=as191896a1 To: sip:08058913...@113.34.235.106 Contact: sip:0345001...@123.50.217.143 Call-ID: 0f3fbfe3463035d04f05534824a18...@113.34.235.106 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 01 Jul 2010 02:20:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 267 v=0 o=root 22702 22702 IN IP4 123.50.217.143 s=session c=IN IP4 123.50.217.143 t=0 0 m=audio 17262 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - Is it possible to retrieve the value of contact in asterisk ? Please let me know. Is there anyone who knows the solution? I need this urgent. Thanks in advance Nahar -- _ -- 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 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
What OS are you running on the two systems? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 10:40 AM, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- 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] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com'
One thing I would do is something like On system one: rpm -qa | sort sys1 On system two: rpm -qa | sort sys2 Then on either system do a diff of these two files. If you only use yum or rpm to install and update software you can tell what is different between the two systems. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 12:23 PM, Faisal Hanif wrote: Both have CentOS 5.2. Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson Sent: Tuesday, June 15, 2010 11:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' What OS are you running on the two systems? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 15, 2010, at 10:40 AM, Faisal Hanif wrote: Till now I am not able to find any difference between both machines. Can you please tell me how I can try to resolve it on OS level using some utility like dig? Regards, Faisal Hanif -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry Miller Sent: Tuesday, June 15, 2010 10:08 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk 1.6.2 WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' On Tue, Jun 15, 2010 at 09:35:37PM +0500, Faisal Hanif wrote: Hi, I am also wonder that same SRV record is working fine on one machine but not on 2nd while both have same asterisk version. It may be some missing OS utilities which asterisk using to resolve SRV? Could be. To test, does replacing whsvoip.globalipcom.com with, say, whs-proxy00.de.whsvoip.globalipcom.com (or whatever is reachable) make it work? What is different about the two machines you've tried? -- Barry -- _ -- 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 -- _ -- 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] get Asterisk version from within dialplan
Starting with version 1.6.x there is a VERSION function that I think will give you the version number. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jun 9, 2010, at 5:19 AM, Vieri wrote: Simple enough: How can I get Asterisk version from within my dialplan? (preferably without calling an AGI script that parses asterisk -rx show version) Is it available as a global variable? Vieri -- _ -- 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 message problems - retransmit and lost messages
I have an asterisk system in Costa Rica that connects to a SIP provider in Atlanta. Sometimes SIP packets seem get dropped or retransmitted too quickly. In trying to debug this I turned on SIP debug in Asterisk and the SIP provider enabled packet capture on his end. What I saw was me sending an invite, them sending a 100 Trying, me sending a cancel, me sending a retransmit of the cancel, me sending another retransmit of the cancel, them sending a 200 ok, them sending a 488 Not Acceptable Here, and then me sending an ACK. What they saw was the invite from me, them sending the trying, me sending two cancels, them sending an ok, me sending a cancel and them sending an ok. I have two questions. First is can I change the time Asterisk waits before doing the retransmit? Second is they do not take down the call because I guess the sequence of packets is not acceptable to them so we end up with hundreds of calls that Asterisk thinks have been dealt with but the provider still thinks are pending some action. Can something be done from the Asterisk side to deal with this situation? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Sangoma Call Progress Analysis behind NAT router
I work at home with standard residential cable Internet service and I wanted to test CPA for use with our dialer solution. The first problem I ran into is that CPA only works with a SIP provider that does IP based authentication opposed to usename/password authentication. After I got an account setup to solve that problem I thought I was on my way to being able to test. No so. I got asterisk making outbound calls via the SIP provider. I got CPA installed and ready to go. I made a test call and although the call gets setup and I can hear audio in both directions CPA did not have any audio to analyze. I then looked at both the SIP messages asterisk sent and received as well as the SIP messages that CPA sent and received and saw that the invite message has the internal IP address for where RTP traffic is to be sent in the CPA messages. inline: CPA.jpg This diagram from CPA's user manual sure looks like my setup. I contacted Sangoma support and they say the product does not do NAT transversal. Has anyone found a work around this problem? -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.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] run extensions after call moved to queue and answered by member
Which version of asterisk are you running? Older versions allowed for an AGI to be called when the queued call got connected with an agent. core show application queue Queue(queuename[|options[|URL][|announceoverride][|timeout][|AGI]) The optional AGI parameter will setup an AGI script to be executed on the calling party's channel once they are connected to a queue member. Newer versions allow for either an AGI or a macro. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 20, 2010, at 4:47 AM, Vasiliy G Tolstov wrote: Hello. Can You provide example, how can i run specific extension after incoming call going into queue and answered (but not hangup). (i want to use System(echo .) after member of specific queue answered a call); Thank You. -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- 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] DTMF Input from the User
Use read application -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 19, 2010, at 9:42 AM, taimur hasan wrote: Hello I am new to Asterisk. I want to know is there any way to get DTMF input from the user in the Dialplan. Regards Taimur Hasan -THQ- !!!ONE Hotmail: Powerful Free email with security by Microsoft. Get it now. -- _ -- 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] Sip phone does not call
The two phones belong to context phones and the two extensions are in context internal. In context phones you need to include = internal so that context phones knows about those extensions. Or put the two extensions in context phones and not context internal. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 19, 2010, at 2:05 PM, ayodele abejide wrote: Hello group, I have asterisk running on my ubuntu machine, and I have a peer to peer network with an XP machine, both of the running x-lite client, I try calling either of the soft phone from the other and the response I get is on my asterisk console is as below: [May 19 19:31:18] NOTICE[1476]: chan_sip.c:20006 handle_request_invite: Call from '1000' to extension '3000' rejected because extension not found. [May 19 19:31:39] NOTICE[1476]: chan_sip.c:21298 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1000 [May 19 19:31:44] NOTICE[1476]: chan_sip.c:20006 handle_request_invite: Call from '1000' to extension '1000' rejected because extension not found. My Diaplan Settings (extensions.conf) [globals] [general] autofallthrough=yes [default] exten = s,1,Verbose(1|Unrouted call handler) exten = s,n,Answer() exten = s,n,Wait(1) exten = s,n,Playback(tt-weasels) exten = s,n,Hangup() [incoming_calls] [internal] exten = 1000,1,Verbose(1|Extension 1000) exten = 1000,n,Dial(SIP/1000,30) exten = 1000,n,Hangup() exten = 3000,1,Verbose(1|Extension 3000) exten = 3000,n,Dial(SIP/1000,30) exten = 3000,n,Hangup() Sip Settings (sip.conf) [general] context=default bindport=5060 srvlookup=yes [1000] type=friend host=dynamic context=phones [3000] type=friend host=dynamic context=phones Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. -- _ -- 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] Agents
Here is what I do to handle agent login/logout
; Agent login logout
exten = *20,1,Verbose(2,Doing agent login/logout)
exten = *20,n,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = *20,n,GotoIf($[${UserID}=]?NOUSER)
exten = *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten = *20,n,GotoIf($[${AgentStatus}=1]?VERIFY)
exten = *20,n,GotoIf($[${AgentStatus}=2]?VERIFY)
exten = *20,n(NOUSER),Playback(cfmc/bad-agent)
exten = *20,n,Playback(vm-goodbye)
exten = *20,n,Hangup()
exten = *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten = *20,n,GotoIf($[${AgentStatus}=2]?AGENTOFF)
exten = *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten = *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten =
*20,n,AddQueueMember(support,Local/queue${agentnumb...@ansqueue${CUT(CHANNEL,-,1)})
; AQMSTATUS can be ADDED | MEMBERALREADY | NOSUCHQUEUE
exten = *20,n,Playback(agent-loginok)
exten = *20,n,Verbose(2,Agent ${AgentNumber} added
${DB(users/${UserID}/AgentDevice)})
exten = *20,n,Hangup()
exten = *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten = *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten = *20,n,RemoveQueueMember(support,Local/queue${agentnumb...@ansqueue)
exten = *20,n,Playback(agent-loggedoff)
exten = *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten = *20,n,Hangup()
[ansqueue]
exten = _Queue.,1,Set(AgentNumber=${EXTEN:5})
exten = _Queue.,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = _Queue.,n,Set(AgentDevice=${DB(users/${UserID}/AgentDevice)})
exten = _Queue.,n,Verbose(2,Agent ${AgentNumber} status is
${DEVSTATE(${AgentDevice})})
exten = _Queue.,n,GotoIf($[${DEVSTATE(${AgentDevice})}=NOT_INUSE]?DIALIT)
exten = _Queue.,n,Busy()
exten = _Queue.,n,Hangup()
exten = _Queue.,n(DIALIT),Dial(${AgentDevice},,g)
exten = _Queue.,n,Hangup()
[support]
exten = 201,1,Verbose(2,Doing support call)
exten = 201,n,Answer()
exten = 201,n,Wait(0.5)
exten = 201,n,Set(qac=${QUEUE_MEMBER_COUNT(support)})
exten = 201,n,GotoIf($[${qac} 0]?HAVEAGNT)
exten = 201,n,Verbose(2,No agents free in support queue)
exten = 201,n,Playback(cfmc/support-no-agent)
exten = 201,n,Voicemail(2...@ourvm,u)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
exten = 201,n(HAVEAGNT),Playback(cfmc/support-intro)
exten = 201,n,Verbose(2,Queuing caller for support agent)
exten = 201,n,Queue(support,nrt,,,120)
exten = 201,n,Verbose(2,Support agent did not answer call)
exten = 201,n,Voicemail(2...@ourvm,b)
exten = 201,n,Playback(goodbye)
exten = 201,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 17, 2010, at 5:30 AM, Peter Childs wrote:
On 17 May 2010 08:40, Lenz Emilitri lenz.lo...@gmail.com wrote:
Use Addmember and removemeber instead :)
l.
Hmm I'm getting that kind of.
From What I can work out.
Agents have been deprecated and are going to be removed.
The replacement, is some complex dialplan using Local Channels which
the admin will have to dream up for themselves.
I'm quite happy to use some new method, but I don't really understand
how yet as all the docs I can find point to using agents
Ideally I need to be able to
a Log into a queue, both by dialing and using the management API
AgentCallbackLogin
b Log Out a que, both by dialing and using the management API
System(agent logoff agent/x) or agentlogoff in management api.
c If the SIP channel (Phone) is not working (Unavailable) remove it
from the queue.
autologoffunavail=yes in agents.conf (but it don't seam to work)
d If the phone is not answered within 10 secs log remove it from the que..
autologoff=10 in agent.conf
e Allow hotdesking extensions so that people don't always need to
login to the same extension.
dial(agent/${EXTEN})
f If the queue is empty or nobody is handling the que drop out, and
ring every phone.
joinempty=strict, leavewhenempty=strict
Using Asterisk 1.4 and a Sark 850.
Any help, or at least where to go
Peter.
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Re: [asterisk-users] play a sound file directly to a caller channel
We do the following:
Action: Originate
Channel: Local/do_playb...@cfmc_cdi_private
Exten: do_chanspy
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=PlayBack
Variable: CfMC_WhatToPlay=lyrics-louie-louie
Variable: CfMC_WhoHear=SIP/GXP280_18-0002
ActionID: PlayBack
Async: true
exten = do_playback,1,Answer()
exten = do_playback,n,UserEvent(BeforePlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_playback,n,Wait(0.3)
exten = do_playback,n,Playback(${CfMC_WhatToPlay})
; PLAYBACKSTATUS - SUCCESS FAILED
exten = do_playback,n,UserEvent(AfterPlayBack,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear}
${PLAYBACKSTATUS})
exten = do_playback,n,Hangup()
exten = do_chanspy,1,Answer()
exten = do_chanspy,n,UserEvent(BeforeChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,ChanSpy(${CfMC_WhoHear},qW)
exten = do_chanspy,n,UserEvent(AfterChanSpy,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_WhatToPlay} ${CfMC_WhoHear})
exten = do_chanspy,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 16, 2010, at 4:16 AM, Daniel Knoll wrote:
Hello User-List,
is it possible to play a sound file directly to a caller channel?
Like this AMI command
Action: Originate
Channel: SIP/20-1d41
Application: Playback
Data: /path/to/audio/file
I get an Error Message. My intension is to play a sound file to a caller and
the other callers don't hear this.
Can someone help me ?
Thanks a lot
Bye Daniel
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Re: [asterisk-users] OK, I'm stumped
Use a dialplan to do what you want and dial that. Originate a call to the first person and point it at context, exten, priority that plays the sound file and then does a dial command to the number you want them to talk to. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 16, 2010, at 7:15 AM, Bruce Ferrell wrote: I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell -- _ -- 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] Are there AMI commands to manipulate a voice mailbox?
You might be able to use local channels to do what you want. As for the user asterisk runs as and the user the web server run as you can maybe have both users belong to the same secondary group and gain the access you need that way. Partly depends on what exactly you are wanting to do. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On May 13, 2010, at 5:09 PM, Carlos Chavez wrote: I want to make a web interface so my users can listen/erase voicemails. Is there a way to do this from the Asterisk manager interface? Since Asterisk and the web server do not run as the same user I cannot do a direct manipulation of the voicemail files in /var/spool/asterisk/voicemail. Maybe there are some AMI commands to delete a specific voicemail from a mailbox? I have not found any so far but documentation is often behind of implementation. Any ideas on how to approach this? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- 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] More clarification on outbound sip channels.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 10, 2010, at 7:35 AM, Eddie Mikell wrote:
Jim, and all:
Thanks for the response.
If I can repeat what you are saying: you don't have to define the multiple
lines in sip.conf?
Your SIP provider will limit the number of concurrent outbound calls you can
make. If you try to dial more than allowed you will get a SIP message with some
error indicating all outbound channels in use.
For comparison, with my current esi setup, we have 10 outgoing lines. If one
line is busy, then the service just rolls to the next number. This is set up
with the phone service.
That doesn't have to done with outgoing sip lines? Does the dialstatus have
to be checked when a user dials out?
SIP calls set setup by talking to your SIP provider. They take care of limiting
concurrency. Both inbound and outbound. You can have logic in your dialplan
using functions GROUP and GROUP_COUNT to keep track of how many channels you
are using. Doing this allows you to play a sound file saying all lines are busy
try your call later.
If the dial command fails then ${DIALSTATUS} will have values like CHANUNAVAIL
CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL TORTURE INVALIDARGS
I understand the incoming lines - we will have a block of DID numbers, and I
can check those and send to appropriate extensions.
Thanks all for helping to clarify. I have gotten a couple of users who
haven't been able to call out, and wasn't sure if I wasn't rolling over the
sip lines properly.
Best,
Eddie Mikell
From: Jim Dickensondicken...@cfmc.com
Subject: Re: [asterisk-users] Multiple SIP lines.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:eda8102c-b255-46e0-940d-1ef217566...@cfmc.com
Content-Type: text/plain; charset=us-ascii
I think it is typical to have some limited number of outbound channels to
your SIP provider. You send all calls, up to your limit, to the same place.
The phone numbers your provider gave you are used to route inbound calls to
your asterisk box. You will typically have some limited number of inbound
channels. All people could call the same number, again controlled by the
number of channels your provider allows. A reason to have multiple inbound
(DID) numbers is so you can route each number to a specific dialplan
extension. You might route one number to the CEO of the company and the other
to a voice tree that allows the caller to specify the person's extension they
want to talk with.
-- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On
May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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Re: [asterisk-users] Multiple SIP lines.
I think it is typical to have some limited number of outbound channels to your
SIP provider. You send all calls, up to your limit, to the same place. The
phone numbers your provider gave you are used to route inbound calls to your
asterisk box. You will typically have some limited number of inbound channels.
All people could call the same number, again controlled by the number of
channels your provider allows. A reason to have multiple inbound (DID) numbers
is so you can route each number to a specific dialplan extension. You might
route one number to the CEO of the company and the other to a voice tree that
allows the caller to specify the person's extension they want to talk with.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On May 7, 2010, at 11:17 AM, Eddie Mikell wrote:
All:
Still experimenting with the asterisk server for the company.
My local phone company has given me two sip numbers to experiment with,
say 444-456-1234 444-456-5678
Calling in and out works, and I've spread a couple of the phones out
with some co-workers.
My question is this: Do I have to define multiple sip lines in either
the sip.conf or the extensions.conf?
Now when I dial out, I just use
exten = _9.,1,DIAL(SIP/${EXTEN:1...@xx.tracfone.net).
How does it know which sip channel to use?
Hope that is clear.
Thanks for all the help.
Eddie Mikell
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Re: [asterisk-users] Issue with (pattern) matching extension
I banged my head with a like problem a few days ago.
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
n does not mean the letter n in a pattern it has a special meaning!
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 29, 2010, at 1:33 AM, Ishfaq Malik wrote:
Philip A. Prindeville wrote:
Here's a segment of my dialplan, I'm working on the freenum/ISN
functionality:
same = n,Set(isnresult=${ENUMLOOKUP(${EXTEN},sip,,1,freenum.org)})
same = n,GotoIf($[${isnresult} != ]?:fn-CONGESTION,1)
; set up our outgoing call state
same = n,Set(SIPFROMUSER=${CALLERID(num)})
same = n,GotoIf($[${GLOBAL(FREENUMDOMAIN)} == ]?dial:)
same = n,Set(SIPFROMDOMAIN=${GLOBAL(FREENUMDOMAIN)})
same = n(dial),Dial(SIP/${isnresult},40)
same = n,Goto(fn-${DIALSTATUS},1)
exten = fn-BUSY,1,Busy()
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
same = n,Congestion()
and the logging:
== ast_get_enum(num='555*9', tech='sip', suffix='freenum.org',
options='', record=1
== ENUM options(): pos=1, options='2'
== ISN ENUM: left=555, middle='9.'
== ast_get_enum() profiling: FAIL, 5.5.5.9.freenum.org, 21 ms
-- Executing [555*99...@outbound-freenum2:5] Set(SIP/guest_1-0010,
isnresult=) in new stack
-- Executing [555*99...@outbound-freenum2:6]
GotoIf(SIP/guest_1-0010, 0?:fn-CONGESTION,1) in new stack
-- Goto (outbound-freenum2,fn-CONGESTION,1)
[Apr 28 16:55:22] WARNING[5987]: pbx.c:4358 __ast_pbx_run: Channel
'SIP/guest_1-0010' sent into invalid extension 'fn-CONGESTION' in
context 'outbound-freenum2', but no invalid handler
pbx*CLI
Note that the string fn-CONGESTION isn't matching the extension pattern:
exten = _fn-.,1,NoOp(ISN: ${DIALSTATUS})
and I'm not sure why.
Anyone want to venture how to go about figuring out how?
Hi
Try
exten = _fn-[A-Z].,1,NoOp(ISN: ${DIALSTATUS})
Ish
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Re: [asterisk-users] dialplan
Are talking about something like exten = _..,1,Noop(Have in this extension) There is also this function that can be used to look for sub strings inside a string. core show function REGEX -= Info about function 'REGEX' =- [Syntax] REGEX(regular expression data) [Synopsis] Regular Expression [Description] Returns 1 if data matches regular expression, or 0 otherwise. Please note that the space following the double quotes separating the regex from the data is optional and if present, is skipped. If a space is desired at the beginning of the data, then put two spaces there; the second will not be skipped. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 28, 2010, at 5:49 AM, wassim darwich wrote: Hi guys: i need to set an extension in my dialplan in which it divert calls if the extension contain specific series ,For example : I need to divert calls which contain to specific extension (contain ,not start or end with), as i know i should set Gotoif command but i dont know what to set after that,Any help will be appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial plan question.
Do you mean you want
exten = bob,1,Dial(SIP/ext-sip/${EXTEN},20)
You want to call out via sip user ext-sip to that system's extension bob?
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On Apr 28, 2010, at 10:50 AM, Aditya Kumar wrote:
Thanks Steve, I corrected spelling that but still having issue :-)
Issue:
when some one calls bob, I want asterisk to add @DOMAIN and make the call.
but it is not working .
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Config files:
sip.conf
[ext-sip]
type=friend
context=phones
qualify=yes
host=external.proxy.com
extensions.conf
exten = bob,1,Dial(SIP/${ext...@ext-sip,20)
the call is not working,
log says:
chan_sip.c:5344 create_addr:no such host: ext-sip
app_dail.c:1745 unable to create channel of type 'SIP' (cause 20-unknown)
can u please correct me what I am missing
From: Steve Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wed, April 28, 2010 12:57:54 AM
Subject: Re: [asterisk-users] Dial plan question.
On 28 Apr 2010, at 06:53, Aditya Kumar wrote:
exten = bob,1,Dial(SIP/${exte...@ext-sip,20)
Where did you define EXTERN?
S
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Re: [asterisk-users] dialplan question
In your sip.conf your permit line does not have an ip address to allow the register from so the call is coming in as a guest and that is likely using context default. Set the permit line to either the ip address of the phone or the network the phone is on. permit=192.168.1.0/255.255.255.0 with allow from any 192.168.1.x address as an example. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 4:31 AM, Vasiliy G Tolstov wrote: Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong? extensions.conf: [office] exten = 601,1,Answer() exten = 601,2,Wait,2 exten = 601,3,Dial(SIP/601,20) exten = 601,4,Hangup() exten = 500,1,Answer() exten = 500,2,Wait,2 exten = 500,3,Dial(SIP/500,20) exten = 500,4,Hangup() sip.conf: [601] deny=0.0.0.0/0.0.0.0 context=office type=friend secret=601 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/601 canreinvite=no callgroup=1 pickupgroup=1 callerid=device 601 accountcode= call-limit=50 [500] deny=0.0.0.0/0.0.0.0 username=500 context=office type=friend secret=500 qualify=yes ;port=5060 permit=0.0.0.0/0.0.0.0 nat=no mailbox=...@device host=dynamic dtmfmode=rfc2833 dial=SIP/500 canreinvite=no callgroup=1 pickupgroup=1 callerid=device 500 accountcode= call-limit=50 -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- 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] dialplan question
I am not sure what you are asking here. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 27, 2010, at 6:16 AM, Vasiliy G Tolstov wrote: В Втр, 27/04/2010 в 06:11 -0700, Jim Dickenson пишет: In your sip.conf your permit line does not have an ip address to allow the register from so the call is coming in as a guest and that is likely using context default. Set the permit line to either the ip address of the phone or the network the phone is on. permit=192.168.1.0/255.255.255.0 with allow from any 192.168.1.x address as an example. Thank You. But a get work with this lines: exten = 102,1,Answer() exten = 102,2,Dial(SIP/102,20) exten = 102,3,Hangup() exten = 500,1,Answer() exten = 500,2,Dial(SIP/500,20) exten = 500,3,Hangup() exten = 601,1,Answer() exten = 601,2,Dial(SIP/601,20) exten = 601,3,Hangup() -- Vasiliy G Tolstov v.tols...@selfip.ru Selfip.Ru -- _ -- 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
