[asterisk-users] Function transfer RFC 5589
Hello, I have the following scenario: 1. VoIP Gateway G400 connected to PSTN 2. Asterisk server 1 (working as IVR) 3. Asterisk server 2 (working as ACD, with several agents connected) I have incoming calls coming from PSTN through the VoIP Gateway to Asterisk server 1 (IVR). When the IVR ends working with the call, transfers it to the Asterisk server 2 (ACD). In Asterisk server 1 (IVR) I'm using the function Transfer() http://www.voip-info.org/wiki/view/Asterisk+cmd+Transfer, which sends a SIP REFER to the VoIP Gateway, in the way that is explained in RFC 5589 http://tools.ietf.org/html/rfc5589#section-6.1 Where: VoIP Gateway G400 is de Transferee Asterisk server 1 (IVR) is the Transferor And Asterisk server 2 (ACD) is the Transfer Target The SIP transaction is completed correctly, with the difference that there is no INVITE (hold) from Transferor to Transferee. In the G400 once finalized the transaction there is no audio: I think is a problem in the G400 because I have done the same test with a Vega gateway and with a softphone and the call is transferred correctly (also audio). 1) Does any one know if exists any problem with transferences, in this gateway? By other side, I'd like to do an attended transfer as is explained in the same RFC 5589 http://tools.ietf.org/html/rfc5589#page-24 2) Is it possible to do that with Transfer function? 3) There is another way (different to use transfer function) to do this kind of transferences? Thanks in advance -- *David Pinedo García* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] Early media recognition
Finally I could do it using the AMI Originate https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerAction_Originate command and the parameter EarlyMedia=true. So, when you throw the call, when detects the EarlyMedia (SIP 183) the channels is bridged to context and you can do the recording. This works from Asterisk 11. On Fri, Jun 27, 2014 at 11:00 AM, David Pinedo dpin...@presenceco.com wrote: Hello, Throwing calls from Asterisk to PSTN (via a VoIP gateway) some operators sends an explaining audio, in situations as: The phone number does is not assigned The phone is powered off etc. The audio is sent before the call to be answered. So, in an automatic dialling application I'd like to recognize that audio to know what to do with those calls (queue them to a service, mark as wrong number, unavailable, etc.). Does Asterisk have any functionality to recognize the audio in early media (similar to the answer machine detection)? Is there a way to record the audio in early media to implement my own early media detector? Thank you in advance --- David Pinedo -- *PRESENCE TECHNOLOGY* *David Pinedo García* Software Developer C/ Comte d'Urgell 240 3º-A Barcelona 08036 dpin...@presenceco.com Ph: +34 93 10 10 300/313 *www.presenceco.com* http://www.presenceco.com/ *Follow us on:* *Presence Technology - Disclaimer* This message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted. For additional information, please visit our website *www.presenceco.com http://www.presenceco.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] CDR(dst) not set in AEL macro
Hi
Probably are this bug, my Asterisk version is 1.8.15.0(the version of
report is 1.8.15.1).
I see that the problem occurs when the call is answered, if busy, fail,
unanswered, the field dst is correct. I see too that when the problem
occurs the dcontext field is set as name of the macro(aparently, the return
in macro no works).
I will go try with others versions and report the status.
Thank's.
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
2014-07-12 7:33 GMT-03:00 Johan Wilfer li...@jttech.se:
2014-07-11 15:38, Rafael dos Santos Saraiva skrev:
Hi
I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field CDR(dst), showing only ~~s~~.
I tried various configurations, but without solutions.
This is the macro:
macro dial-out(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} = 1) {
Set(__PICKUPMARK=${destno});
if(${ODBC_verify_user(${CALLERID(num)})} 0) {
t = tT;
} else {
t = t;
}
} else {
t = T;
}
Dial(${dialstring}/${destno},30,${t});
return;
}
I don't know if this is maybe related to this:
https://issues.asterisk.org/jira/browse/ASTERISK-20441
If it is this is a bug in the AEL compiler I think.
--
Johan Wilfer
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[asterisk-users] R: Asterisk and Call Hold
Hi All, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value a=sendrecv is present, according to the rfc3264 the sdp value a must be mark with sendonly. I've already tried with Asterisk 1.8 and Asterisk 11, but there is the same problem. I've already read all the information about canreinvite and directmedia Can anybody help me? Thanks a lot Marco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PJSIP outbound register and inbound calls
Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600@192.168.1.1:5060 client_uri=sip:600@192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all allow=ulaw outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 match=192.168.1.1 when I call 600 from other pbx I getting an notice NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from 'Ilya sip:502@192.168.1.1' failed for '192.168.1.1:5060' (callid: ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint found and Not Accessable on phone let's imagine that 600 its external number of voip operator, and I wanna accept all incoming calls from it (no matter what caller id it has) what I doing wrong? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every 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] PJSIP outbound register and inbound calls
Nick Awesome wrote: Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600@192.168.1.1:5060 client_uri=sip:600@192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all allow=ulaw outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 match=192.168.1.1 when I call 600 from other pbx I getting an notice NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from 'Ilyasip:502@192.168.1.1' failed for '192.168.1.1:5060' (callid: ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint found and Not Accessable on phone let's imagine that 600 its external number of voip operator, and I wanna accept all incoming calls from it (no matter what caller id it has) what I doing wrong? When receiving calls from a VoIP provider you have to match using the source IP address. You also don't authenticate as the provider will refuse to do so. When you control both ends it's really up to you whether to do the matching based on the source IP address OR use a user account with authentication. If using the user account the user portion of the From header has to be set to the username (from_user in pjsip, fromuser in chan_sip). Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] R: Asterisk and Call Hold
Marco Colombo wrote: Hi All, Kia ora, I have a problem with asterisk and call hold. In the re-invite package when I take the call to the hold, the SDP value “a=sendrecv” is present, according to the rfc3264 the sdp value a must be mark with “sendonly”. Are you referring to a call being put on hold? If so this is correct. Internally the musiconhold just becomes a different source of audio, the fact it is on hold does not get reflected out the SIP signaling. People have mentioned they'd like this (as well as being able to passthrough a hold request) but nobody I know of has worked on it. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] PJSIP outbound register and inbound calls
I thought that type=identify will match an IP address and accept it, well, in my example I can control both sides and able to configure it without registration. in real life I have a provider that requires username/password authentication provider gives me - Username - Password - DomainName I have configure it like I showed before and have exactly the same notice [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from 'cb5069 sip:asterisk@85.195.98.178' failed for '85.195.98.178:5060' (callid: 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching endpoint found 85.195.98.178 is an operator, so what I should add to my config to be able accept calls from Registered peer ? On Jul 16, 2014, at 7:55 PM, Joshua Colp jc...@digium.com wrote: Nick Awesome wrote: Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600@192.168.1.1:5060 client_uri=sip:600@192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all allow=ulaw outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 match=192.168.1.1 when I call 600 from other pbx I getting an notice NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from 'Ilyasip:502@192.168.1.1' failed for '192.168.1.1:5060' (callid: ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint found and Not Accessable on phone let's imagine that 600 its external number of voip operator, and I wanna accept all incoming calls from it (no matter what caller id it has) what I doing wrong? When receiving calls from a VoIP provider you have to match using the source IP address. You also don't authenticate as the provider will refuse to do so. When you control both ends it's really up to you whether to do the matching based on the source IP address OR use a user account with authentication. If using the user account the user portion of the From header has to be set to the username (from_user in pjsip, fromuser in chan_sip). Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] PJSIP outbound register and inbound calls
Nick Awesome wrote: I thought that type=identify will match an IP address and accept it, well, in my example I can control both sides and able to configure it without registration. in real life I have a provider that requires username/password authentication provider gives me - Username - Password - DomainName They may require it for *outgoing* calls to them but for incoming I highly doubt they'd want you to authenticate them. It's usually always IP authentication. I have configure it like I showed before and have exactly the same notice [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from 'cb5069sip:asterisk@85.195.98.178' failed for '85.195.98.178:5060' (callid: 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching endpoint found 85.195.98.178 is an operator, so what I should add to my config to be able accept calls from Registered peer ? The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described. Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
Re: [asterisk-users] PJSIP outbound register and inbound calls
Ok there is my test account from sipiko.net username: cb5069 password: sqv664yqtp domain: callme.sipiko.net its using username/password authentication. because its just website widget I need only inbound calls from this peer, test call can be done from url: http://callme.sipiko.net/callme.php?id=5069call_id=210tunnel=yes on my side I have an asterisk 12 using pjsip Have configured IVR with number 5000 on context dialmap, so I need forward all calls from this provider to number 5000 over dialmap context help if you can please:) On Jul 16, 2014, at 8:53 PM, Joshua Colp jc...@digium.com wrote: Nick Awesome wrote: I thought that type=identify will match an IP address and accept it, well, in my example I can control both sides and able to configure it without registration. in real life I have a provider that requires username/password authentication provider gives me - Username - Password - DomainName They may require it for *outgoing* calls to them but for incoming I highly doubt they'd want you to authenticate them. It's usually always IP authentication. I have configure it like I showed before and have exactly the same notice [Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from 'cb5069sip:asterisk@85.195.98.178' failed for '85.195.98.178:5060' (callid: 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching endpoint found 85.195.98.178 is an operator, so what I should add to my config to be able accept calls from Registered peer ? The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described. Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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
[asterisk-users] Simultaneous Ring
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial another set of
numbers while continuing to ring, or interrupting the first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
Thanks,
Scott Haley
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Re: [asterisk-users] Simultaneous Ring
asterisk-users-boun...@lists.digium.com wrote on 07/16/2014 01:46:09 PM:
From: Haley,Scott A scott.ha...@edwardjones.com
To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com,
Date: 07/16/2014 01:46 PM
Subject: [asterisk-users] Simultaneous Ring
Sent by: asterisk-users-boun...@lists.digium.com
I have a need to issue a dial command to a number:
same = n,Dial(${DIALGROUP1},${TIMER1},t)
After a number of seconds, let's say 10 seconds. I want to dial
another set of numbers while continuing to ring, or interrupting the
first group of numbers.
same = n,Dial(${DIALGROUP2},${TIMER1},t)
Is there a way to do this without interrupting the first call?
Thanks,
Scott Haley
I believe that what you want to do is best done with Local Channels. See
this link:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
for more information.--
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Re: [asterisk-users] Simultaneous Ring
After a number of seconds, let's say 10 seconds. I want to dial another set of numbers while continuing to ring, or interrupting the first group of numbers. I'd use queues. First set of numbers in queue 1, both the first set and second set of number in queue 2. If the call falls out of queue 1, it should fall into queue 2. Then use the ringall strategy on both queues Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
