Re: [asterisk-users] Dahdi Answer a Call On ringing State.
Hi Richard, i update a new version of asterisk to 1.8.9.1 and checked the issue are still same and my call getting answer while it is in ringing. here is brief details for finding root cause. Dahdi -Version: 2.4.1.2 Echo Canceller: OSLEC[channels] File : chan_dahdi.conf context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no usecallerid=yes callerid=asreceived cidstart=polarity_in cidsignalling=dtmf callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callprogress=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 relaxdtmf=yes pulsedial=yes ;Uncomment these lines if you have problems with the disconection of your analog lines busydetect=yes busycount=3 immediate=no answeronpolarityswitch=yes polarityonanswerdelay=1000 group=0 channel = 1 group=1 channel = 2 group=0 channel = 3 group=0 channel = 4 Let me know your thoughts on this thanks Dhaval On Thu, Feb 16, 2012 at 1:47 PM, Richard Mudgett rmudg...@digium.comwrote: I have setup Dahdi with Sangoma FXO A200 card and asterisk 1.8 , everything seems fine and working perfectly incoing/outgoing. but one major issue is, when i made an out call from dahdi trunks and when a number is in ringing state it gives me an answer state. This was recently fixed by https://issues.asterisk.org/jira/browse/ASTERISK-18841 so i cannot develop any custom application which can use a screening macro because when a cellphone is in ringing state call is answered by dahdi channel so it will start executing dial plan which causes an issue. 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] Dahdi Answer a Call On ringing State.
FXO ports are considered Answered as soon as dialing completes. This is the way analog FXO ports work. Use PRI or SIP if you need correct Answer supervision. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, February 16, 2012 6:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State. Hi Richard, i update a new version of asterisk to 1.8.9.1 and checked the issue are still same and my call getting answer while it is in ringing. here is brief details for finding root cause. Dahdi -Version: 2.4.1.2 Echo Canceller: OSLEC[channels] File : chan_dahdi.conf context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no usecallerid=yes callerid=asreceived cidstart=polarity_in cidsignalling=dtmf callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callprogress=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 relaxdtmf=yes pulsedial=yes ;Uncomment these lines if you have problems with the disconection of your analog lines busydetect=yes busycount=3 immediate=no answeronpolarityswitch=yes polarityonanswerdelay=1000 group=0 channel = 1 group=1 channel = 2 group=0 channel = 3 group=0 channel = 4 Let me know your thoughts on this thanks Dhaval On Thu, Feb 16, 2012 at 1:47 PM, Richard Mudgett rmudg...@digium.com wrote: I have setup Dahdi with Sangoma FXO A200 card and asterisk 1.8 , everything seems fine and working perfectly incoing/outgoing. but one major issue is, when i made an out call from dahdi trunks and when a number is in ringing state it gives me an answer state. This was recently fixed by https://issues.asterisk.org/jira/browse/ASTERISK-18841 so i cannot develop any custom application which can use a screening macro because when a cellphone is in ringing state call is answered by dahdi channel so it will start executing dial plan which causes an issue. 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
[asterisk-users] How to receive SMS ?
Hi, I've read here and there how Asterisk could send SMS but I didn't find much about how to receive SMS and forward them to an email box. 1. First of all, I don't think my telco would let me receive any SMS my landline. 2. Maybe I could find providers selling this service for a monthly fee; 3. I could build and operate my own infrastructure. Given this asterisk-users mailing-list purpose, and for curiosity's sake, how could I build my own SMS reception service with Asterisk (1.6.1 or later) ? Which channel (chan_mobile, chan_datacard, ...) and hardware would be appropriate ? Suggestions ? 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
Re: [asterisk-users] Asterisk + Avaya (CM5.2) H.323 trunk Link
I have Avaya IPOffice 403 talking to my Asterisk 1.8.x with virtually no issues using OOH323. I am having some minor issues with the name portion of the caller ID sent to Avaya. That may be relalted to a way FreePBX created the dial plan. Maybe not. Never had time to systematically look into this one. I spent some time working with the OOH323 maintainer / developer to iron out the networking issues specific to Avaya's implementation of H.323. -Vladimir On 2/14/2012 1:56 PM, Dustin fails wrote: Anyone have an H.323 trunk tied between their Avaya and Asterisk box that works? I am having some issues trying to get the two systems to connect. I am using the ooh323 channel to try to make the connection between the two system. I have all my configs if anyone would like to look over them. If I do a trace on Avaya I get a denial event 1191: Network Failure. 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] How to receive SMS ?
Le 16/02/2012 16:55, Olivier a écrit : Hi, I've read here and there how Asterisk could send SMS but I didn't find much about how to receive SMS and forward them to an email box. 1. First of all, I don't think my telco would let me receive any SMS my landline. Why? If I assume well you're in France, so no problem. 2. Maybe I could find providers selling this service for a monthly fee; If your point 1) view is true, it will change nothing ;-) 3. I could build and operate my own infrastructure. What we did. Anyway, the problem is not here. If you follow http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it's working, but, at least for France (true around one year back): - SFR is sending SMS to landlines as voice messages which is completely stupid as they ask to press 1 to listen the message or you can wait 15 sec and start recording, perhaps you will get it or part of. But if they change their announce message, you have at first to know it and then recalculate the delay. Crazy. An answer machine will have same problem. - FREE mobile doesn't send SMS to landlines - ORANGE is working but we also faced some SMS sended as voice message - BOUYGUES wasn't tested We stopped to work on this as each mobile operator do what he want, SMS gateways being not better. To unstable to rely on it. Given this asterisk-users mailing-list purpose, and for curiosity's sake, how could I build my own SMS reception service with Asterisk (1.6.1 or later) ? Which channel (chan_mobile, chan_datacard, ...) and hardware would be appropriate ? Suggestions ? Hope that helped -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 receive SMS ?
2012/2/16, Administrator TOOTAI ad...@tootai.net: Le 16/02/2012 16:55, Olivier a écrit : Hi, I've read here and there how Asterisk could send SMS but I didn't find much about how to receive SMS and forward them to an email box. 1. First of all, I don't think my telco would let me receive any SMS my landline. I meant my telco wouldn't let me receive any SMS Why? If I assume well you're in France, so no problem. You mean you can receive SMS on a landline in France (or the opposite) ? 2. Maybe I could find providers selling this service for a monthly fee; If your point 1) view is true, it will change nothing ;-) 3. I could build and operate my own infrastructure. What we did. Anyway, the problem is not here. If you follow http://www.voip-info.org/wiki/view/Asterisk+cmd+Sms it's working, but, at least for France (true around one year back): - SFR is sending SMS to landlines as voice messages which is completely stupid as they ask to press 1 to listen the message or you can wait 15 sec and start recording, perhaps you will get it or part of. But if they change their announce message, you have at first to know it and then recalculate the delay. Crazy. An answer machine will have same problem. - FREE mobile doesn't send SMS to landlines - ORANGE is working but we also faced some SMS sended as voice message - BOUYGUES wasn't tested We stopped to work on this as each mobile operator do what he want, SMS gateways being not better. To unstable to rely on it. If a gateway has its own SIM card and GSM stuff, should it receive SMS ? Given this asterisk-users mailing-list purpose, and for curiosity's sake, how could I build my own SMS reception service with Asterisk (1.6.1 or later) ? Which channel (chan_mobile, chan_datacard, ...) and hardware would be appropriate ? Suggestions ? Hope that helped Yes it helped ! -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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?
https://reviewboard.asterisk.org/r/1599/ I so wish that this patch would be backported to the 1.8 branch! I am considering switching to trunk just for this alone. I know it's a stretch but, given the popularity of running Fail2Ban alongside Asterisk, could it not fall under the pretense of 'security risk' that someone very easily breaks Fail2Ban by forgetting to set verbose back to 5 during a routine CLI session where they might have temporarily needed to reduce verbosity? I know I'm reaching, but doesn't hurt to beg. Luke -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Saturday, December 31, 2011 5:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that? On Fri, Dec 30, 2011 at 09:03:34PM -0500, Bruce B wrote: So, based on what you are saying if I issue the command core set verbose 0 and then exit the system Fail2Ban will stop working for Asterisk (this is since Fail2ban works based on the log file entries). Can anyone else please confirm that as well. Though in trunk you can set different log levels to different files. -- Tzafrir Cohen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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?
A definite me too from my side. Always wondered why it wasn't like that. It would do wonders to help us fix our own problems instead of filling in bugs or posting here ;-) (hint hint) Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Luke Hamburg Sent: Thursday, February 16, 2012 2:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that? https://reviewboard.asterisk.org/r/1599/ I so wish that this patch would be backported to the 1.8 branch! I am considering switching to trunk just for this alone. I know it's a stretch but, given the popularity of running Fail2Ban alongside Asterisk, could it not fall under the pretense of 'security risk' that someone very easily breaks Fail2Ban by forgetting to set verbose back to 5 during a routine CLI session where they might have temporarily needed to reduce verbosity? I know I'm reaching, but doesn't hurt to beg. Luke -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Saturday, December 31, 2011 5:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that? On Fri, Dec 30, 2011 at 09:03:34PM -0500, Bruce B wrote: So, based on what you are saying if I issue the command core set verbose 0 and then exit the system Fail2Ban will stop working for Asterisk (this is since Fail2ban works based on the log file entries). Can anyone else please confirm that as well. Though in trunk you can set different log levels to different files. -- Tzafrir Cohen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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?
On 16-02-12 20:18, Luke Hamburg wrote: https://reviewboard.asterisk.org/r/1599/ I so wish that this patch would be backported to the 1.8 branch! I am considering switching to trunk just for this alone. I know it's a stretch but, given the popularity of running Fail2Ban alongside Asterisk, could it not fall under the pretense of 'security risk' that someone very easily breaks Fail2Ban by forgetting to set verbose back to 5 during a routine CLI session where they might have temporarily needed to reduce verbosity? I know I'm reaching, but doesn't hurt to beg. Yes that would be a very nice addition. Perhaps someone with Asterisk coding skills can backport the patch. Have you checked if it applies at all to the latest 1.8 master? I wonder if that patch is already part of 10 master or the 10.2 branch as I could not see anything mentioned on reviewboard. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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?
- Original Message - From: Patrick Lists asterisk-l...@puzzled.xs4all.nl To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 16, 2012 6:32:13 PM Subject: Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that? On 16-02-12 20:18, Luke Hamburg wrote: https://reviewboard.asterisk.org/r/1599/ I so wish that this patch would be backported to the 1.8 branch! I am considering switching to trunk just for this alone. I know it's a stretch but, given the popularity of running Fail2Ban alongside Asterisk, could it not fall under the pretense of 'security risk' that someone very easily breaks Fail2Ban by forgetting to set verbose back to 5 during a routine CLI session where they might have temporarily needed to reduce verbosity? I know I'm reaching, but doesn't hurt to beg. Yes that would be a very nice addition. Perhaps someone with Asterisk coding skills can backport the patch. Have you checked if it applies at all to the latest 1.8 master? I wonder if that patch is already part of 10 master or the 10.2 branch as I could not see anything mentioned on reviewboard. It's not in Asterisk 10, it's in the current trunk, which will eventually become Asterisk 11. The patch, while a very nice and useful enhancement, is unfortunately fairly intrusive. I can't see it becoming part of the Asterisk 1.8 or Asterisk 10 branches, given (a) the fact that it is certainly an improvement and not a bug fix, and (b) the risk involved in back-porting a patch of that magnitude and scope. Regards, Patrick Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Set(CALLERID(name)) when incoming call is anonymous
Hi,
I'm trying to figure out why I can't pass through caller ID details
that I set manually if the incoming call that I am forwarding was
anonymous.
Our reception staff need to know which number the client was calling
in on so they can give the right greeting message when answering.
E.g. I have the following in our dialplan for one reception number
(similar for others):
G_RECEPTION=SIP/SIP/
exten = 12345678,1,Set(CALLERID(name)=ORG1)
exten = 12345678,n,Set(CALLERID(name-pres)=allowed)
exten = 12345678,n,Dial(${G_RECEPTION},15,i)
exten = 12345678,n,VoiceMail(12345678,su)
exten = 12345678,n,Hangup()
Normally this works great with the name ORG1 and the client's number
both appearing on the handset (Snom 320). However, if the caller had
no caller ID this shows up on the screen as Anonymous.
Setting name-pres actually doesn't have noticable effect. I added that
later when trying to find a solution.
How can I make ORG1 show up on the screen when the caller has no
caller ID?
TIA,
Kevin Shanahan
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Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that?
Fair enough. Giving up on the backport to 1.8 or 10 for now, I had a thought for a kludge. How about a shell script (scheduled with cron) that checks for any 'active' consoles -- any connected consoles where there has been user input within the last X minutes. If it finds none, then set the verbosity back to 5 (or whatever level you want). There are a few problems with this -- I couldn't find any way to: 1) query Asterisk for a count or list of console connections, much less 'active' ones 2) query Asterisk for the current verbosity level (without changing it) Am I barking up another wrong tree here? Anyone have any other ideas on how to solve this problem? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matthew Jordan Sent: Thursday, February 16, 2012 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] High verbose set at console effects the logger file Full - Why is that? It's not in Asterisk 10, it's in the current trunk, which will eventually become Asterisk 11. The patch, while a very nice and useful enhancement, is unfortunately fairly intrusive. I can't see it becoming part of the Asterisk 1.8 or Asterisk 10 branches, given (a) the fact that it is certainly an improvement and not a bug fix, and (b) the risk involved in back-porting a patch of that magnitude and scope. Matthew Jordan Digium, Inc. | Software Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Set(CALLERID(name)) when incoming call is anonymous
Kevin,
You might have luck changing the callerid number so its not empty, that might
override the Anonymous label.
exten = 12345678,1,GotoIf,$[${LEN(${CALLERID(num)})} != 0]?3
exten = 12345678,2,Set(CALLERID(num)=0)
exten = 12345678,3, Your code starts here
Good luck!
Mark
On Feb 16, 2012, at 9:26 PM, Kevin Shanahan wrote:
Hi,
I'm trying to figure out why I can't pass through caller ID details
that I set manually if the incoming call that I am forwarding was
anonymous.
Our reception staff need to know which number the client was calling
in on so they can give the right greeting message when answering.
E.g. I have the following in our dialplan for one reception number
(similar for others):
G_RECEPTION=SIP/SIP/
exten = 12345678,1,Set(CALLERID(name)=ORG1)
exten = 12345678,n,Set(CALLERID(name-pres)=allowed)
exten = 12345678,n,Dial(${G_RECEPTION},15,i)
exten = 12345678,n,VoiceMail(12345678,su)
exten = 12345678,n,Hangup()
Normally this works great with the name ORG1 and the client's number
both appearing on the handset (Snom 320). However, if the caller had
no caller ID this shows up on the screen as Anonymous.
Setting name-pres actually doesn't have noticable effect. I added that
later when trying to find a solution.
How can I make ORG1 show up on the screen when the caller has no
caller ID?
TIA,
Kevin Shanahan
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Re: [asterisk-users] Set(CALLERID(name)) when incoming call is anonymous
On Fri, Feb 17, 2012 at 12:56:15PM +1030, Kevin Shanahan wrote:
I'm trying to figure out why I can't pass through caller ID details
that I set manually if the incoming call that I am forwarding was
anonymous.
Our reception staff need to know which number the client was calling
in on so they can give the right greeting message when answering.
E.g. I have the following in our dialplan for one reception number
(similar for others):
G_RECEPTION=SIP/SIP/
exten = 12345678,1,Set(CALLERID(name)=ORG1)
exten = 12345678,n,Set(CALLERID(name-pres)=allowed)
exten = 12345678,n,Dial(${G_RECEPTION},15,i)
exten = 12345678,n,VoiceMail(12345678,su)
exten = 12345678,n,Hangup()
Normally this works great with the name ORG1 and the client's number
both appearing on the handset (Snom 320). However, if the caller had
no caller ID this shows up on the screen as Anonymous.
Okay, it seems to work if I add:
exten = 12345678,n,Set(CALLERID(num-pres)=allowed)
I guess if either of Name or Number aren't allowed then Asterisk
replaces the whole caller id string with the default:
Anonymous sip:Anonymous@anonymous.invalid
Cheers,
Kevin Shanahan
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Re: [asterisk-users] Asterisk RTCP
Hello, Thanks for taking out tome for my query. Yes I do have an actual problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers port mirrored to it). My end points(soft-phones) are sending RTCP connection strings to asterisk, and Asterisk then forwards their call to their destination choosing any suitable carrier. If I don't get RTCP flowing through asterisk the monitoring tool simply fails to display and call stats. Please advice what should I be doing to cater this. Thanks, Sammy On Thu, Feb 16, 2012 at 10:00 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/16/2012 01:16 AM, Sammy Govind wrote: Hello list, I need to know about Asterisk's friendly nature with RTCP. I've phones which support RTCP and they connect to the outer world via multiple carriers. In one of my recent packet traces I've observed that the caller initiated a call with rtcp string in SDP while for the same call dialling our from Asterisk to the carrier has no RTCP string in SDP ! Can anyone please tell why is this so! or if there is anything I can do to make RTCPs flow through the asterisk server ! I've asterisk 1.6.2.20 in production. It is not mandatory to signal anything related to RTCP in the SDP. RTCP is implicitly handled on the next port up from the port being used for RTP; the signaling in SDP is only needed if the RTCP is *not* going to be on the next port up. RTCP will never *flow through* Asterisk, as Asterisk is terminating both RTP flows and thus is an endpoint for both of them. Do you have an actual problem you are trying to resolve, or are you just asking questions about RTCP? -- 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-**usershttp://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] Dahdi Answer a Call On ringing State.
Hi Eric, but in this case dialing is not completed ring is still going on, so it should not answered thanks Dhaval On Thu, Feb 16, 2012 at 7:52 PM, Eric Wieling ewiel...@nyigc.com wrote: FXO ports are considered Answered as soon as dialing completes. This is the way analog FXO ports work. Use PRI or SIP if you need correct Answer supervision. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Thursday, February 16, 2012 6:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dahdi Answer a Call On ringing State. Hi Richard, i update a new version of asterisk to 1.8.9.1 and checked the issue are still same and my call getting answer while it is in ringing. here is brief details for finding root cause. Dahdi -Version: 2.4.1.2 Echo Canceller: OSLEC[channels] File : chan_dahdi.conf context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no usecallerid=yes callerid=asreceived cidstart=polarity_in cidsignalling=dtmf callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes callprogress=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 relaxdtmf=yes pulsedial=yes ;Uncomment these lines if you have problems with the disconection of your analog lines busydetect=yes busycount=3 immediate=no answeronpolarityswitch=yes polarityonanswerdelay=1000 group=0 channel = 1 group=1 channel = 2 group=0 channel = 3 group=0 channel = 4 Let me know your thoughts on this thanks Dhaval On Thu, Feb 16, 2012 at 1:47 PM, Richard Mudgett rmudg...@digium.com wrote: I have setup Dahdi with Sangoma FXO A200 card and asterisk 1.8 , everything seems fine and working perfectly incoing/outgoing. but one major issue is, when i made an out call from dahdi trunks and when a number is in ringing state it gives me an answer state. This was recently fixed by https://issues.asterisk.org/jira/browse/ASTERISK-18841 so i cannot develop any custom application which can use a screening macro because when a cellphone is in ringing state call is answered by dahdi channel so it will start executing dial plan which causes an issue. 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
