Re: [asterisk-users] Goggle voice incoming dialplan
On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.comwrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts ( wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
both show transfercapability DIGITAL
Regards
Robb
On 16 June 2011 23:40, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup
this still dials and aswers in 1.8 but no frames are passed and the
call times out and drops
I have also tried
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
with exactly the same outcome,
Both of these methods should work after doing a quick look a the code.
Does the outgoing call SETUP indicate digital capability?
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every 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] Goggle voice incoming dialplan
Could you elaborate on how you can associate those non-gmail accounts with gchat account? On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.comwrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts ( wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 Log in Mysql
On Thu, 2011-06-16 at 19:12 -0300, Henrique Fernandes wrote: It is possible to log queue in mysql without turning on realtime asterisk? Thanks! []'sf.rique -- Hi Yes, you can pick and choose which things you want to use your DB by defining them in your extconfig.conf so, in extconfig.conf you would need to add queue_log=mysql,your-db-name,queue_log in res_config_mysql.conf (1.8) or res_mysql.conf (1.4,1.6) you would have to put in the connection details for your database If you are using 1.8 your table create statement would be CREATE TABLE `queue_log` ( `id` int(10) unsigned NOT NULL auto_increment, `time` char(26) default NULL, `callid` varchar(32) NOT NULL default '', `queuename` varchar(32) NOT NULL default '', `agent` varchar(32) NOT NULL default '', `event` varchar(32) NOT NULL default '', `data` varchar(100) NOT NULL default '', `data1` VARCHAR(100), `data2` VARCHAR(100), `data3` VARCHAR(100), `data4` VARCHAR(100), `data5` VARCHAR(100), PRIMARY KEY (`id`) )ENGINE=InnoDB ; Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Missed calls and groups
Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten = 200,1,Dial(Sip/112SIP/113SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the specific phone I do want displayed so just disabling the Missed Calls feature on the phone doesn't cut the mustard. Ideas? (I'd also want this to work with Queues but let's see about the basics first) -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Missed calls and groups
You could use the c option in the dial command which sends a call answered elsewhere reason to the phone and then the phone won't record it in the missed list (I know it works on the snom I didn't check it on the yealink ) But you'll have to send that only with the dial command which you don't want recorded Regarding queues if you call agents using the specific channel driver like sip/200 then it works but if using the local channel driver there was a bug reported (as far I remember) that it didn't work (it might have been fixed) -Original Message- From: russ...@lls.lls.com (Russell Brown) Sender: asterisk-users-boun...@lists.digium.com Date: Fri, 17 Jun 2011 12:30:21 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Missed calls and groups Is there a SIP header I can set (for Snom and Yealink phones if that's relevant) or any other mechanism to tell a phone to ignore a particular call from it's missed call list? I have bits of the dialplan that ring groups of phones eg: exten = 200,1,Dial(Sip/112SIP/113SIP/114) and I don't want such calls being recorded by the phone as a missed call. Calls to the specific phone I do want displayed so just disabling the Missed Calls feature on the phone doesn't cut the mustard. Ideas? (I'd also want this to work with Queues but let's see about the basics first) -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Goggle voice incoming dialplan
I have a free google apps account (http://www.google.com/a I think) setup for SelbyTech.com. Basically it is a gmail account, just with a different domain. Thanks, --Warren Selby, dCAP On Jun 17, 2011, at 2:43 AM, asterisk asterisk aster...@ck-lee.com wrote: Could you elaborate on how you can associate those non-gmail accounts with gchat account? On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.com wrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts (wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] RTP Streaming
Hi Users, I would like to know about the RTP audio streaming. I am taking the example as youtube, in youtube if bandwidth is less the application will buffer and will stream the video; likewise how to do with audio buffering and play the file using RTP in asterisk. Any guide of clue will make me understand. Thank you in advance. Thank you, Gopal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Goggle voice incoming dialplan
I'm not trying to be a jerk or anything. But have you played with this at all or are you just looking for someone to write you a dialplan/config that already works. There are some great pointers in the sample configs and if you look around on google for gtalk asterisk. Also, read the asterisk wiki on how the gtalk stuff works. As far as how to tie your gapps account to gtalk that IS outside the scope of this list. -- cobra2 Http://linuxindixie.info Warren Selby wcse...@selbytech.com wrote: I have a free google apps account (http://www.google.com/a I think) setup for SelbyTech.com. Basically it is a gmail account, just with a different domain. Thanks, --Warren Selby, dCAP On Jun 17, 2011, at 2:43 AM, asterisk asterisk aster...@ck-lee.com wrote: Could you elaborate on how you can associate those non-gmail accounts with gchat account? On Fri, Jun 17, 2011 at 2:38 PM, Warren Selby wcse...@selbytech.com wrote: On Thu, Jun 16, 2011 at 10:58 PM, asterisk asterisk aster...@ck-lee.com wrote: Can this non gmail.com GV number be terminated at some sip accounts so that I can bridge to it via asterisk as client? Yes, I've setup some GV numbers on my google apps accounts (@selbytech.com, for example), and associated those with gchat accounts (wcse...@selbytech.com), and successfully received calls on my asterisk using this solution. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] click to call
hello list i need to create a call files in order to do a click to call with asterisk1.4 i want to use sip 223 in order to call phone number i have created a file.call in var/spool/asterisk/tmp and i move it to var/spool/asterisk/outgoing but there is no call please tell me if there is any worng Channel: SIP/223 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: call-file-test Extension: 223 extensions.conf [call-file-test] exten = 223,1,Dial(SIP/223,tT) exten = 223,2,hangup thanks and regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] background audio for inbound leg
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
Re: [asterisk-users] click to call
On Fri, Jun 17, 2011 at 05:20:39PM +, salaheddine elharit wrote: i want to use sip 223 in order to call phone number Is that meant to be the originator or the destination? Channel: gets the originator; Extension: gets the destination. Roger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk voicemail distribution groups
Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 voicemail distribution groups
Or is there anyway to have a message copied from a mailbox to a list of other mailboxes everytime a message is left in it? On Fri, Jun 17, 2011 at 1:54 PM, vip killa vipki...@gmail.com wrote: Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 voicemail distribution groups
On Fri, Jun 17, 2011 at 1:05 PM, vip killa vipki...@gmail.com wrote: Or is there anyway to have a message copied from a mailbox to a list of other mailboxes everytime a message is left in it? On Fri, Jun 17, 2011 at 1:54 PM, vip killa vipki...@gmail.com wrote: Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes To the best of my knowledge there is no way built into asterisk to do this, however, you can easily write a script that does this using your favorite scripting language and have asterisk call that program whenever a voicemail is left using the mailcmd= option in voicemail.conf. This option gives you better control over the entire message delivery process. Another option is the externnotify= command, but that is run on more occasions than just when a voicemail is left. -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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] asterisk voicemail distribution groups
On Fri, 2011-06-17 at 13:54 -0400, vip killa wrote: Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes You can user Voicemail(100101102103) to send the same voicemail to as many people as necessary. The first mailbox listed is the one used for the personal welcome message. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup
this still dials and aswers in 1.8 but no frames are passed and the
call times out and drops
I have also tried
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
with exactly the same outcome,
Both of these methods should work after doing a quick look a the code.
Does the outgoing call SETUP indicate digital capability?
both show transfercapability DIGITAL
Could be a problem in the media stream handling not being setup for digital
mode.
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every 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 voicemail distribution groups
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Friday, June 17, 2011 2:53 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk voicemail distribution groups On Fri, 2011-06-17 at 13:54 -0400, vip killa wrote: Is there any to have asterisk record a file then send that file to a distribution list of voicemail boxes? What I'm trying to accomplish is a prompt for a user to record/listen to their message and then choose to send the recording to multiple voicemail box's inboxes You can user Voicemail(100101102103) to send the same voicemail to as many people as necessary. The first mailbox listed is the one used for the personal welcome message. This is documented in the output from core show application voicemail, and is documented on voip-info.org. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Next Asterisk 1.8 Release
Hi, When is the next release planned for as very keen to get it into Production but require the call pickup fix. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridged Digital call
any reason why this would happen, should I report a bug on the issue
tracker?
Robb
On 17 June 2011 19:55, Richard Mudgett rmudg...@digium.com wrote:
Hi All
Just upgraded from 1.6? to 1.8.4.1
I ised to be able to get a digital call working across a bridged
isdn
channel in 1.6 and 1.4 using the following;-
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,(Set(CHANNEL(transfercapability)=DIGITAL)
exten = _X.,6,dial(DAHDI/g1/${EXTEN})
exten = _X.,7,hangup
this still dials and aswers in 1.8 but no frames are passed and the
call times out and drops
I have also tried
exten = _X.,1,gotoif($[${TRANSFERCAPABILITY}=DIGITAL]?5:)
exten = _X.,2,dial(DAHDI/g1/${EXTEN})
exten = _X.,3,Noop(${CHANNEL})
exten = _X.,4,hangup
exten = _X.,5,Noop
exten = _X.,6,dial(DAHDI/g1d/${EXTEN})
exten = _X.,7,hangup
with exactly the same outcome,
Both of these methods should work after doing a quick look a the code.
Does the outgoing call SETUP indicate digital capability?
both show transfercapability DIGITAL
Could be a problem in the media stream handling not being setup for digital
mode.
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every 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] Bridged Digital call
Just upgraded from 1.6? to 1.8.4.1 I ised to be able to get a digital call working across a bridged isdn channel in 1.6 and 1.4 using the following;- [snip] Could be a problem in the media stream handling not being setup for digital mode. ..., should I report a bug on the issue tracker? Did anything change outside of Asterisk? (Different ISDN equipment or configuration for instance.) If not then yes I think it is a bug since you say it used to work with v1.4 and v1.6.x. I think it could be a problem in the media stream handling not being setup for digital mode. For completeness, the bug report should have attached: 1) chan_dahdi.conf (and any files it includes) 2) Debug capture files of pri set debug on span x output of a call attempt for the incoming call leg and the outgoing call leg. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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-biz] Ground Start ATA / VOIP Gateway
On 2011-06-14 15:51, Robert Huddleston wrote: I only need 4 fxs... I looked at the IAD2431 but it uses T1/E1 as WAN... If I could assign Fast Ethernet to WAN that would be great... Budget is not that great I've done that on a 2431. There's nothing special about the T1 port. I've made it into a fractional PRI and run the fa0/0 as the WAN or added another T1 port and used that as the WAN. Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.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] Polycom BLF
From http://www.voip-info.org/wiki/view/Asterisk+presence
Polycom Phones (updated for 3.2.X firmware with asterisk 1.6.1 Jan/2010) With
SIP 3.2.X firmware (available on the Polycom download site) and Asterisk
1.6.1, Polycom phones now support a full featured BLF showing statuses of
Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension and enable directed pickup. Directed pickup is enabled by adding
the following lines to extensios.conf
exten = _*8.,1,SET(GLOBAL(PICKUPMARK)=${EXTEN:2})
exten = _*8.,n,Pickup(${EXTEN:2}@PICKUPMARK)
On the phone side for each line that is going to be monitored add lines like
the following to the phone's cfg file.
attendant.reg=1
attendant.resourceList.1.address=sip:205@192.168.1.102
attendant.resourceList.1.label=205
attendant.resourceList.2.address=sip:217@192.168.1.102
attendant.resourceList.2.label=217
call.directedCallPickupMethod=legacy
call.directedCallPickupString=*8
feature.12.name=directed-call-pickup
feature.12.enabled=1
Assuming my server is at 192.168.1.102, this will add two BLF lines to the
phone for extensions 205 and 217. Calls incoming to those extensions will
show a blinking green led on the monitoring phone, pressing the hard key
will pick the call up, if it is answered elsewhere the led will change to
solid red. AFAIK this cannot be configured via the phones web gui, you must
use the cfg files. You can also use versions of Asterisk older than 1.6.1 if
you remove the restriction on what asterisk thinks Polycom phones can
handle. Look in chan_sip.c for
if (strstr(p-useragent, Polycom)) {
p-subscribed = XPIDF_XML;
and change that line to
p-subscribed = DIALOG_INFO_XML;
On Tue, Jun 14, 2011 at 8:36 AM, Jeff LaCoursiere j...@sunfone.com wrote:
Struggling with an IP650 and 7 IP335s this morning. I have the following
hints defined (courtesy of FreePBX 2.9):
extensions_additional.conf:**exten = 300,hint,SIP/300
extensions_additional.conf:**exten = 301,hint,SIP/301
extensions_additional.conf:**exten = 302,hint,SIP/302
extensions_additional.conf:**exten = 303,hint,SIP/303
extensions_additional.conf:**exten = 304,hint,SIP/304
extensions_additional.conf:**exten = 305,hint,SIP/305
extensions_additional.conf:**exten = 307,hint,SIP/307
extensions_additional.conf:**exten = 308,hint,SIP/308
extensions_additional.conf:**exten = 322,hint,SIP/322
extensions_additional.conf:**exten = 350,hint,SIP/350
extensions_additional.conf:**exten = 400,hint,SIP/400
The Polycoms are all pulling an XML directory via FTP where each extension
has BW (Buddy Watch) set to 1:
item
lnMehra/ln
fnRay/fn
ct301/ct
sd101/sd
bw1/bw
/item
This all actually works fine, and from the reception phone (the 650) I can
see the status of all the extensions, and if I dig into some menus on the
335 I can see status as well. So I would expect that core show hints
would show '8' for all extensions, but it doesn't:
artha*CLI core show hints
-= Registered Asterisk Dial Plan Hints =-
300@ext-local : SIP/300 State:Idle
Watchers 7
301@ext-local : SIP/301 State:Idle
Watchers 8
302@ext-local : SIP/302 State:Idle
Watchers 8
303@ext-local : SIP/303 State:Idle
Watchers 8
304@ext-local : SIP/304 State:InUse
Watchers 8
305@ext-local : SIP/305 State:Idle
Watchers 7
307@ext-local : SIP/307 State:Idle
Watchers 1
308@ext-local : SIP/308 State:Idle
Watchers 7
350@ext-local : SIP/350 State:Idle
Watchers 1
400@ext-local : SIP/400 State:InUse
Watchers 7
- 11 hints registered
Something seems broken here. And the 650 seems to lose its hint for a
phone once in a while, and report it as unreachable, even though it can
easily make and receive calls from it.
Am I tilting at windmills? Is this really unstable or has someone made it
work solidly?
Thanks!
--
Jeff LaCoursiere
SunFone
340-715-7600 x222
j...@sunfone.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:
Re: [asterisk-users] No audio after a reinvite changing codec
Inviato da iPhone Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling ewiel...@nyigc.com ha scritto: We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. Hi Eric, this behavior is an asterisk bug or asterisk can never change the codec on the fly? Thanks, Matteo -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Thursday, June 16, 2011 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No audio after a reinvite changing codec On 15/06/2011 8:15 PM, Matteo Campana wrote: HI list, no idea?? :) There not much substance in the information provided for an assessment to be made. I would suggest you capture the network traffic between UAC (g711) Asterisk UAS ensuring the snap length is large enough to capture the whole packet and do the same with traffic between Asterisk UAC Provider then use Wireshark and its telephony feature to analyse VoIP calls, check the flows, you may discover the problem this way! Larry. M. On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana matteo.camp...@gmail.com wrote: Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g729 --- g729 rtp rtp After a while, we have the reinvite sent by the SIP provider with g711 in the SDP. So asterisk need to change audio codec from g729 to g711 and correctly we see on debug the following line: Oooh, we need to change our audio formats since our peer supports only g729 and asterisk send back 200 OK to the provider. At this point we have one way rtp audio: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g711 --- g711 rtp rtp So the problem is that UAC does not hear audio at all. Any idea? (Asterisk version: 1.4.33.1). Thanks in advance, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] No audio after a reinvite changing codec
I don't know. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matteo Campana Sent: Friday, June 17, 2011 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No audio after a reinvite changing codec Inviato da iPhone Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling ewiel...@nyigc.com ha scritto: We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. Hi Eric, this behavior is an asterisk bug or asterisk can never change the codec on the fly? Thanks, Matteo -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore Sent: Thursday, June 16, 2011 10:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No audio after a reinvite changing codec On 15/06/2011 8:15 PM, Matteo Campana wrote: HI list, no idea?? :) There not much substance in the information provided for an assessment to be made. I would suggest you capture the network traffic between UAC (g711) Asterisk UAS ensuring the snap length is large enough to capture the whole packet and do the same with traffic between Asterisk UAC Provider then use Wireshark and its telephony feature to analyse VoIP calls, check the flows, you may discover the problem this way! Larry. M. On Mon, Jun 13, 2011 at 6:55 PM, Matteo Campana matteo.camp...@gmail.com wrote: Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g729 --- g729 rtp rtp After a while, we have the reinvite sent by the SIP provider with g711 in the SDP. So asterisk need to change audio codec from g729 to g711 and correctly we see on debug the following line: Oooh, we need to change our audio formats since our peer supports only g729 and asterisk send back 200 OK to the provider. At this point we have one way rtp audio: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 -- g711 | g711 --- g711 rtp rtp So the problem is that UAC does not hear audio at all. Any idea? (Asterisk version: 1.4.33.1). Thanks in advance, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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
[asterisk-users] ooh323 errors while compiling: asterisk 1.8.3 and 1.8.4
Hi All; Please I need a help in the ooh323. First of all, the only way to have h323 working in asterisk 1.8.3 or 1.8.4 is to use ooh323? There is no way to get the normal h323 channel that come with asterisk to work fine !! Now, let us see the ooh323 problem that I am facing: Already I compiled ptlib and h323plus successfully (and I was use them to run gnugk). From make menuselect, when I tried to select chan_ooh323 from Add-ons without selecting ADDON from the Module Embedding, it is compiling and installing without error message. But it look like that not loading well the chan_ooh323 as I did netstat -ln and did not see any thing related to 1720. From the CLI, I unloaded chan_ooh323 and then I loaded chan_ooh323, but it gave me: Call-Bilal*CLI module load chan_ooh323.so Loaded chan_ooh323.so [Jun 17 20:23:32] NOTICE[2392]: chan_ooh323.c:2506 reload_config: Unable to load config ooh323.conf, OOH323 disabled Loaded chan_ooh323.so = (Objective Systems H323 Channel) Again, from make menuselect, if I selected chan_ooh323 from the Add-ons and I selected ADDON from module embedding. Then I ran ./configure and make. I got an error like the following: /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:4259: multiple definition of `onModeChanged' /usr/bin/ld: Dwarf Error: found dwarf version '1024', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xb870): first defined here ../addons/chan_ooh323.o: In function `configure_local_rtp': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:3888: multiple definition of `configure_local_rtp' /usr/bin/ld: Dwarf Error: found dwarf version '5', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0x92f0): first defined here ../addons/chan_ooh323.o: In function `ooh323_set_write_format': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:1413: multiple definition of `ooh323_set_write_format' /usr/bin/ld: Dwarf Error: found dwarf version '116', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xb1f0): first defined here ../addons/chan_ooh323.o: In function `close_udptl_connection': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:4133: multiple definition of `close_udptl_connection' /usr/bin/ld: Dwarf Error: found dwarf version '1026', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xbe90): first defined here ../addons/chan_ooh323.o: In function `ooh323_set_read_format': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:1472: multiple definition of `ooh323_set_read_format' /usr/bin/ld: Dwarf Error: found dwarf version '1570', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xc350): first defined here ../addons/chan_ooh323.o: In function `setup_udptl_connection': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:4074: multiple definition of `setup_udptl_connection' /usr/bin/ld: Dwarf Error: found dwarf version '1800', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xc870): first defined here ../addons/chan_ooh323.o: In function `close_rtp_connection': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:4009: multiple definition of `close_rtp_connection' /usr/bin/ld: Dwarf Error: found dwarf version '6', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xce30): first defined here ../addons/chan_ooh323.o: In function `setup_rtp_connection': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:3976: multiple definition of `setup_rtp_connection' /usr/bin/ld: Dwarf Error: found dwarf version '5893', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xd040): first defined here ../addons/chan_ooh323.o: In function `ooh323_destroy': /usr/src/asterisk-1.8.3.2/addons/chan_ooh323.c:3462: multiple definition of `ooh323_destroy' /usr/bin/ld: Dwarf Error: found dwarf version '512', this reader only handles version 2 information. ../addons/chan_ooh323.eo:(.text+0xe5d0): first defined here collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 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
Re: [asterisk-users] Inbound call not dialing exten
sir its done
thank you A.J
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
On Thu, Jun 16, 2011 at 3:12 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Thursday 16 Jun 2011, mahesh katta wrote:
-- Executing Set(Zap/3-1, Dest=50{EXTEN:-2}) in new
stack
-- Executing MixMonitor(Zap/3-1,
/var/spool/asterisk/astrec/20110616-122003-0559566768-4578924-1308212403.2
225.gsm|av(0)V(0)) in new stack
-- Executing Dial(Zap/3-1, SIP/50{EXTEN:-2}||tTo) in new
stack
Oops, my bad; I missed out a punctuation mark.
Before I tell you the answer, though, have a good look at the console
diagnostic messages and see if you can spot for yourself what it's doing
wrong. What is it trying to dial, and what *should* it be trying to dial?
And didn't it feel good, knowing you fixed it yourself?
-- SPOILER FOLLOWS --
There should be a $ sign in the Set() step:
Set(Dest=50${EXTEN:-2})
so it will use the rightmost 2 characters of ${EXTEN}.
--
AJS
Answers come *after* questions.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every 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] No audio after a reinvite changing codec
On 18/06/2011 5:36 AM, Matteo Campana wrote: Inviato da iPhone Il giorno 16/giu/2011, alle ore 16:37, Eric Wielingewiel...@nyigc.com ha scritto: We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. Hi Eric, this behavior is an asterisk bug or asterisk can never change the codec on the fly? Thanks, Matteo The problem reported occurs after a fax tone is detected, one might expect T.38 or G711 to be used to handle the fax, not G729! There is no configuration file information for each of the nodes/peers, no debug of each peer involved nor a trace of the packets hence no one will have ideas! Larry. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
