Re: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
Am 27.09.2014 17:28, schrieb d tbsky: can someone give an example for the function? thanks for the help. Not a programmer here, just grep -r'ed through the code, but maybe try one of these: G711A G711_ALAW -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Intercom Telephone Feature
Dear all, My client has Asterisk based telephony system. He needs to add the intercom feature in his telephones. He has 300 concurrent users with two PRI Channels. I want to check if there is a possibility to have the requested scenario by adding this feature to his current telephone system I would appreciate your help so much. Best Wishes, Dania Abu Asi Sales Executive Engineer Future Trends Establishment Abu Dhabi - U.A.E. Mob : +971 50 4948363 Off : +971 2 6730666 Fax : +971 2 6734888 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Intercom Telephone Feature
Of course, it is possible. Depending on what the desired behavior is, it might suffice to enable the auto-answer feature of an end point. You might also want to read about paging and intercom for different scenarios. jg Dear all, My client has Asterisk based telephony system. He needs to add the intercom feature in his telephones. He has 300 concurrent users with two PRI Channels. I want to check if there is a possibility to have the requested scenario by adding this feature to his current telephone system -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
2014-09-28 14:01 GMT+08:00 Markus unive...@truemetal.org: Am 27.09.2014 17:28, schrieb d tbsky: can someone give an example for the function? thanks for the help. Not a programmer here, just grep -r'ed through the code, but maybe try one of these: G711A G711_ALAW thanks a lot for help!! I tried both but none works. maybe this function can not work like the old channel variable SIP_CODEC, which can change inbound call codec. but I do notice something different between chan_sip and chan_pjsip. I use zoiper softphone for testing: when I dialout sip trunk with chan_sip, the remote peer rings, and zoiper now shows what codec to use. if I use SIP_CODEC before dial to change the codec, zoiper will use the new CODEC, but asterisk internal won't change and still transcoding in the middle.(at least core show channel sip/x told me transcoding) when I dialout sip trunk with chan_pjsip, the remote peer rings, but zoiper didn't show what codec to use. only after the callee answer the phone, zoiper shows what codec to use. so it seems chan_pjsip have better chance to do the right thing without transcoding. it's sad that chan_pjsip won't select best codec match two peers automatically without transcoding. but I hope it at least can provide a magic function or channel variable like SIP_CODEC/SIP_CODEC_INBOUND to make correct codec selection. Regards, tbskyd -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 append the recording file.
How about recording the call calling it whatever you want, and then using a
custom AGI script to append the call to the original one? That’s how I would do
it if it were me.
Regards;
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anurag Rana
Sent: Sunday, September 28, 2014 1:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to append the recording file.
Hi All,
I am trying to record the call using MixMonitor.
exten=_,n,MixMonitor(${EXTEN}.wav,b)
What i want to do is-
when first time a call is made to some number say 1100, a new file (1100.wav)
is created.
When call is made 2nd or 3rd time, no new file is created instead call
recording is appended to file created in above step.
Now I know that 'a' option is used to append the recording to a file but I
couldn't find any example on how to use it?
Also if I use 'a' option and file doesn't exist then is it created or it is
error?
Any suggestions please?
Anurag Rana
http://newbie42.blogspot.in/
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[asterisk-users] how to make voip client cannot use same username?
Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) -- ok then in client 3, i used (1001) same username with client 1. when client 1 is connecting with client 2, my client 3 could make a call to with client 2 (1002) with the same username in client 1. how i could make the system, so i cannot use with 1 username in 2 different client before i make a call (when registering process in voip client), or at least my voip client cannot use same username if that username is connected with the other user? any help will help me a lot. thanks in advance. rafa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 append the recording file.
On Sun, 28 Sep 2014, Anurag Rana wrote: I am trying to record the call using MixMonitor. ... Now I know that 'a' option is used to append the recording to a file but I couldn't find any example on how to use it? Also if I use 'a' option and file doesn't exist then is it created or it is error? Any suggestions please? Sure. Try it -- faster than waiting for a response. If it depends on the file already existing, add it to 'core show application mixmonitor.' If it creates the file if it doesn't exist, add it to 'core show application mixmonitor.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to append the recording file.
As the other posters said - try it! Another option would be to use sox to combine files with some common part of their filename. On 28 September 2014 19:39, Steve Edwards asterisk@sedwards.com wrote: On Sun, 28 Sep 2014, Anurag Rana wrote: I am trying to record the call using MixMonitor. ... Now I know that 'a' option is used to append the recording to a file but I couldn't find any example on how to use it? Also if I use 'a' option and file doesn't exist then is it created or it is error? Any suggestions please? Sure. Try it -- faster than waiting for a response. If it depends on the file already existing, add it to 'core show application mixmonitor.' If it creates the file if it doesn't exist, add it to 'core show application mixmonitor.' -- 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] Ports leak
check your ulimits :) On 26 September 2014 17:15, CDR vene...@gmail.com wrote: I am using Asterisk 12 svn, from today, and after a few thousand calls, I run out of ports. This happens eith PJSIOP and regular old SIP. I think it is RTP related. Any idea how can I troblshoot this. It happened teh same with Asterisk 11. On the other end there is a freeswitch. My guess is that there is an incompatibility. Thanks in advance for your thoughts -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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 make voip client cannot use same username?
hi, anyone seen this? actually this is my sip.conf file [1002] type = friend context = test username = 1002 secret = 12345 host = dynamic if i want to make my client register to server with matching on username instead of ip address, so the username is used just for 1 client how could i do that? if i change the host (dynamic into the ip) and the type (friend) is still the same. in my voip client is unregistered (however i still could make a call but can't be called). but if the type i changed into peer, my voip client is still unregistered (i couldn't make a call and can't be called). so what i have to configured in my sip.conf and extensions.conf files? thanks in advance. rafa On Sun, Sep 28, 2014 at 10:51 PM, rafa alfurqan rafa.alfur...@gmail.com wrote: Hi All, I have one asterisks server and 3 client (i'm using voip sip client for my handphone). I've configured sip.conf and extension.conf with 3 user different. And nothing wrong with them, i could make them to make a call too. what i want to ask is, i was try to use 1 user (ex:1001) in 2 different client. example: client 1 (1001) make a call to client 2 (1002) -- ok then in client 3, i used (1001) same username with client 1. when client 1 is connecting with client 2, my client 3 could make a call to with client 2 (1002) with the same username in client 1. how i could make the system, so i cannot use with 1 username in 2 different client before i make a call (when registering process in voip client), or at least my voip client cannot use same username if that username is connected with the other user? any help will help me a lot. thanks in advance. rafa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
