Re: [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value uri.
07.03.2015 1:21, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com mailto:serov@gmail.com wrote: 07.03.2015 0:24, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com mailto:serov@gmail.com wrote: Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com http://sip.example.com client_uri=sip:1234567...@sip.example.com mailto:client_uri=sip:1234567...@sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option contact_user. Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov - I believe what you are looking for is already available. See the identify type (type=identify) section that is in the pjsip.conf file and the identify option for endpoints. These allow you to identify and endpoint by IP address. For more information see the pjsip.conf.sample file. Also take a look at configuring Asterisk for res_pjsip [1] specifically the part about configuring endpoint identification by IP address [2]. If you run into problems more information can also be found in the res_pjsip troubleshooting guide [3], specifically the section on identify by IP address [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, Thank you for answer. But... ones again: I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? Simple Example. I have two trunks with their own credentials (and did) to the same sip server: - for home - for bussiness [home-example.com-endpoint] [bussiness-example.com-endpoint] [home-example.com-registration] contact_user=home-example.com-endpoint [bussiness-example.com-registration] contact_user=bussiness-example.com-endpoint ;and ok... i wrote identify by IP section [example.com-identify] type=identify match=example.com http://example.com endpoint= ??? It is very! important for me to know what trunk passes through the incoming call: home or bussiness. 1. Identify by IP. Do you have answer? 2. Identify by username. What? I can't make endpoints to all of my contacts. Ok. I can use contact_user in registraction and route incoming call by INVITE uri. Can i? Dmitriy Serov I don't think I fully understand the scenario, but if you have different named endpoints originating from the same address (or even different addresses) then these can be identified by the username portion of the sip uri. However, if you have endpoints for instance with the same name, but different addresses then these can be distinguished by using the identity type. Your scenario looks like the first option. The endpoint names are different, but the address is the same, so identification based on the username should be sufficient. However, if you have a mix of both types on your system, for instance multiple endpoints on two different systems (IP addresses) with the same names, then I am unsure how you would select the correct endpoint even while attempting to identify_by uri. -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:http://digium.com http://asterisk.org I have different named endpoints originating from the same address. then these can be identified by the username portion of the sip uri ok. A little more examples. case 1. Device! we have: endpoint in config and registered device. incoming call. In this case identification makes by username portion of sip
Re: [asterisk-users] New Asterisk build
Ira wrote: Hello John, Friday, March 6, 2015, 12:34:42 PM, you wrote: Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your But given that means buying an old computer, why change at all? I already have a very low power one that works fine. Is AstLinux better than Centos 5 running Asterisk 13? -- Ira Better? Depends on how you define better Since you haven't revealed what you are currently using, really hard to say, but running a box without a spinning hard drive and fans to die, certainly is better An OS that fits in 1Gig might very well be better than a bloated CentOS 5, 6 or what have you If what you have works for you, then why even ask? If it works, leave it alone. You will certainly find 1000 opinions on the list, if any decide to take the bait JN -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk build
John I will have to get one of these and give this a try. Thanks for sharing. Thanks Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.) 616-855-1030 Ext. 2003 From: John Novack SCII jnov...@stromberg-carlson.org Sent: Friday, March 6, 2015 3:37 PM To: Ira i...@extrasensory.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New Asterisk build Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your older version of Asterisk to the fairly current version 11 currently available with AstLinux. Use the GUI to edit and mage the system, as AstLinux has a somewhat different directory structure than you may be familiar with You should be up and running in a couple of hours, have a low power 20 watts, fanless flash based system that will just work in a real case. The Pi is OK for a playtoy and some testing, but I much prefer the HP thin clients for a robust installation. I assume you are not doing any fancy call center or heavy database work. For a home or home office it is a really good solution. AstLinux is also used with other embedded installations on computers with multiple Ethernet ports, acting as router and firewall in addition. I prefer the component solution personally, which makes the HP thin clients the way to go. John Novack I have built more than 30 of these systems on various HP Thin Clients, used off of eBay with no failures Ira wrote: Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put it on a Raspberry Pi or something like that, but I need the one POTS line and all I have for that at the moment is a Digium card for a PCI slot. Are there any current thoughts on this? -- Ira -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk build
Iran For the kind of loads and low cost you are talking with 2 FXO, 2FXS and SIP the Grandstream UMC6102 is low power feature rich and easy to maintain. Check it out - http://www.grandstream.com/index.php/products/ip-voice-telephony/ip-pbx-solu tions/ucm61xx If you do choose to use the UMC61xx the grandstream phones auto-provision with it well, but it works with any complaint SIP phone. If you do want to go with an asterisk home brew. You could use a Grandstream GXW4104 (4 FXO) for your POTS line. It is a FXO gateway that would register as a SIP endpoint. (You could look at the HT503 which has one FXO port, but I find them to be less reliable then the GXW4104). The nice thing about using gateways is there are no drivers to load on your asterisk build as the gateway is just a SIP endpoint. I have built asterisk test systems on raspberry pi Rev B and have not been happy with their performance even in light loads. The new version 2 B looks like it might be better, In ether case the Gateways would be a good way to go to connect your lines. Watch your SD card speeds slow cards really gave me a lot of issues. Especially when you had someone leaving a voicemail and someone else was trying to listing to an IVR prompt, multiple users reading and writing at the same time just really have not worked well. We hooked up a SSD via USB and put our prompts and voicemail on it and it was a bit better still limited to USB2 speeds, but that increased the cost. The UMC6102 is the best value as buy the time you purchase a gateway, system and spend time loading it is hard to beat the price point and you can get support on it from Grandstream or a reseller. (To be open I am a Grandstream reseller, I am offering these recommending as they are good options. There are several other low cost asterisk like PBX's out there as well, Allo and several others, but I know the GS options work) Good Luck and I hope this info helps. Thanks Bryant Zimmerman (Grand Dial Communications, a ZK Tech Inc.) P.S. Glen's post also offers some good points as well. -- _ -- 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 configuration for AWS/EC2 based Asterisk 13.1.0
BTW, the allow=!all is equivalent to disallow=all, so you can drop the disallow line. On Thu, Mar 5, 2015 at 7:26 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: OK. I think I found the issue. The key is to add rtp_symmetric=yes Here's what my final configuration looks like: [transport-udp] type=transport protocol=udp bind=0.0.0.0 ;; for within EC2 local_net=172.31.32.0/20 ;; For softphones within EC2 local_net=192.168.1.0/24 external_media_address=publicIPOfEC2Instance external_signaling_address=publicIPOfEC2Instance ;Templates for the necessary config sections [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=!all,ulaw direct_media=no rtp_symmetric=yes On Thu, Mar 5, 2015 at 5:52 PM, Sonny Rajagopalan sonny.rajagopa...@gmail.com wrote: Hello All, I have an Asterisk server v13.1.0 running on EC2 and I am able to connect and register SIP devices and see them on the asterisk CLI. I am also able to place calls, but I am not able to hear any audio on either end after the call is picked up. I was wondering if you can tell me what a minimal configuration for Asterisk on EC2 looks like. My current pjsip.conf configuration looks like this: type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 ; In the following two lines, replace publicIP with the output of ; curl -s http://169.254.169.254/latest/meta-data/public-ipv4 external_media_address=publicIP external_signaling_address=publicIP [endpoint_internal](!) type=endpoint context=from-internal disallow=all allow=ulaw direct_media=no [auth_userpass](!) type=auth auth_type=userpass [aor_dynamic](!) type=aor max_contacts=1 remove_existing=yes ;Definitions for our phones, using the templates above ;; usernames and passwords etc. below My security group configuration allows TCP, UDP posrt 5060 inbound, outbound from/to 0.0.0.0/0 and TCP, UDP ports 1-2 from/to 0.0.0.0/0. Should I turn on STUN for my zoiper softphones? Any specific flavor? What am I doing wrong? Any help appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 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
Re: [asterisk-users] Music on hold
Kris Stark wrote: Unfortunately, I was never asked about this to enough detail to be able to tell them how to set up the music, and as a result I have an eight minute file with several different messages all tied together into that one file. You can use Audacity to break that file into multiples http://audacity.sourceforge.net/ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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] Guidence in DialPlan programming.
I am dealing with a FreePBX generated dialplan. I have been following
the processing traces attempting to make use of the advice I received
here respecting setting a custom ring tone. I have discovered that
the context I am using for incoming calls is not used at all during a
blind transfer. Thus setting a third ring tone for that situation
inside that context is an impossibility.
I know now what I need to do and possibly where I need put it. What I
wish is some guidance on how to properly return from my custom code
without damaging the dialplan elsewhere.
Here is the situation:
In extensions.conf I see this:
;-
;-
; Internal dialplan that most internal phones have access to
;
[from-internal]
include = from-internal-noxfer
include = from-internal-xfer
include = bad-number ; auto-generated
;-
;-
; from-internal-noxfer:
;
; Place to put internal dialplan that should not be accessible
; during a blind transfer, this context will not be visible
; during such.
;
[from-internal-noxfer]
include = from-internal-noxfer-custom
include = from-internal-noxfer-additional ; auto-generated
;-
;-
; from-internal-xfer:
;
; Place to put most internal dialplan, will be visible during
; normal calls and blind transfers.
;
[from-internal-xfer]
include = from-internal-custom
include = from-internal-additional ; auto-generated
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
;-
;-
If a call is placed by a local extension then context
[from-internal-noxfer] is used. If a blind transfer is performed the
context is [from-internal-xfer]. What I am considering is placing the
following code in extensions-custom.conf:
[from-internal-custom].
exten = _X,1,Noop()
exten = _X,n,Set(AlertSnom=http://www.notused.com\;info=)
exten = _X,n,Set(AlertInternalTransfer=alert_internal_transfer)
exten = _X,n,Set(__ALERT_INFO=${AlertSnom}${AlertInternalTransfer})
exten = s,1,Macro(hangupcall)
exten = h,1,Macro(hangupcall)
My question is: are the last two lines the correct method of returning
from this back to extensions.conf? Is there something else I should
use? At them moment I just want to know how to properly and safely
return to the original referring context ([from-internal-xfer]).
--
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[asterisk-users] AWS/EC2 server selection
Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that server with ssd is required as all 500+ calls needs to be recorded. Regards, Amit Patkar-- _ -- 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] AWS/EC2 server selection
Why use Amazon? With that kind of load I would want dedicated servers. Call Rackspace or Softlayer. j On 03/06/2015 11:59 AM, Amit Patkar wrote: Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that server with ssd is required as all 500+ calls needs to be recorded. Regards, Amit Patkar -- _ -- 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] cant get incoming calls in asterisk
*friends help me * *cant get incoming calls in asterisk* *(when i connect **80081 in softphone ---every thing is ok**)* *--- SIP read from UDP:200.152.125.221:5060 http://200.152.125.221:5060 ---* *INVITE sip:80081@10.47.10.10:5060 http://sip:80081@10.47.10.10:5060 SIP/2.0* *Record-Route: sip:200.152.125.221;lr;ftag=as6872d065* *Via: SIP/2.0/UDP 200.152.125.221;branch=z9hG4bKd4fd.b3489837.0* *Via: SIP/2.0/UDP 200.152.125.213:5060;branch=z9hG4bK2214551d;rport=5060* *From: 8008 21982008200 sip:111...@ser.sipcode.com.br sip%3a111...@ser.sipcode.com.br;tag=as6872d065* *To: sip:80...@ser.sipcode.com.br sip%3a80...@ser.sipcode.com.br* *Contact: sip:11@200.152.125.213 sip%3A11@200.152.125.213* *Call-ID: 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br* *CSeq: 105 INVITE* *User-Agent: FPBX-2.9.0(1.4.41)* *Max-Forwards: 69* *Date: Fri, 06 Mar 2015 18:17:21 GMT* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO* *Supported: replaces* *Content-Type: application/sdp* *Content-Length: 338* *v=0* *o=root 3211 3214 IN IP4 200.152.125.213* *s=session* *c=IN IP4 200.152.125.213* *t=0 0* *m=audio 14686 RTP/AVP 0 8 3 18 101* *a=rtpmap:0 PCMU/8000* *a=rtpmap:8 PCMA/8000* *a=rtpmap:3 GSM/8000* *a=rtpmap:18 G729/8000* *a=fmtp:18 annexb=no* *a=rtpmap:101 telephone-event/8000* *a=fmtp:101 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* *a=sendrecv* *-* *--- (16 headers 16 lines) ---* *Sending to 200.152.125.221:5060 http://200.152.125.221:5060 (no NAT)* *Using INVITE request as basis request - 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br* *Found peer '11' for '11' from 200.152.125.221:5060 http://200.152.125.221:5060* *--- Reliably Transmitting (no NAT) to 200.152.125.221:5060 http://200.152.125.221:5060 ---* *SIP/2.0 401 Unauthorized* *Via: SIP/2.0/UDP 200.152.125.221;branch=z9hG4bKd4fd.b3489837.0;received=200.152.125.221* *Via: SIP/2.0/UDP 200.152.125.213:5060;branch=z9hG4bK2214551d;rport=5060* *From: 8008 21982008200 sip:111...@ser.sipcode.com.br sip%3a111...@ser.sipcode.com.br;tag=as6872d065* *To: sip:80...@ser.sipcode.com.br sip%3a80...@ser.sipcode.com.br;tag=as09849411* *Call-ID: 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br 5c385d117f894c0f2dd79a3f2129b...@ser.sipcode.com.br* *CSeq: 105 INVITE* *Server: FPBX-12.0.42(11.14.1)* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE* *Supported: replaces, timer* *WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=63fdf36b* *Content-Length: 0* ** -- Best regards Antony моб (066) 919-75-33 моб (063) 656-43-40 satski...@gmail.com mail%3asatski...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Asterisk build
Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put it on a Raspberry Pi or something like that, but I need the one POTS line and all I have for that at the moment is a Digium card for a PCI slot. Are there any current thoughts on this? -- Ira -- _ -- 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] res_pjsip endpoint config object's 'identify_by' option needs new value uri.
Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com client_uri=sip:1234567...@sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option contact_user. Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov -- _ -- 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] res_pjsip endpoint config object's 'identify_by' option needs new value uri.
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com wrote: Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.comclient_uri=sip:1234567...@sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option contact_user. Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov - I believe what you are looking for is already available. See the identify type (type=identify) section that is in the pjsip.conf file and the identify option for endpoints. These allow you to identify and endpoint by IP address. For more information see the pjsip.conf.sample file. Also take a look at configuring Asterisk for res_pjsip [1] specifically the part about configuring endpoint identification by IP address [2]. If you run into problems more information can also be found in the res_pjsip troubleshooting guide [3], specifically the section on identify by IP address [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, -- Kevin Harwell 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
Re: [asterisk-users] res_pjsip endpoint config object's 'identify_by' option needs new value uri.
07.03.2015 0:24, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com mailto:serov@gmail.com wrote: Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com http://sip.example.com client_uri=sip:1234567...@sip.example.com mailto:client_uri=sip:1234567...@sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option contact_user. Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov - I believe what you are looking for is already available. See the identify type (type=identify) section that is in the pjsip.conf file and the identify option for endpoints. These allow you to identify and endpoint by IP address. For more information see the pjsip.conf.sample file. Also take a look at configuring Asterisk for res_pjsip [1] specifically the part about configuring endpoint identification by IP address [2]. If you run into problems more information can also be found in the res_pjsip troubleshooting guide [3], specifically the section on identify by IP address [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, -- Kevin Harwell Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at:http://digium.com http://asterisk.org Thank you for answer. But... ones again: I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? Simple Example. I have two trunks with their own credentials (and did) to the same sip server: - for home - for bussiness [home-example.com-endpoint] [bussiness-example.com-endpoint] [home-example.com-registration] contact_user=home-example.com-endpoint [bussiness-example.com-registration] contact_user=bussiness-example.com-endpoint ;and ok... i wrote identify by IP section [example.com-identify] type=identify match=example.com endpoint= ??? It is very! important for me to know what trunk passes through the incoming call: home or bussiness. 1. Identify by IP. Do you have answer? 2. Identify by username. What? I can't make endpoints to all of my contacts. Ok. I can use contact_user in registraction and route incoming call by INVITE uri. Can i? Dmitriy Serov -- _ -- 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] New Asterisk build
Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your older version of Asterisk to the fairly current version 11 currently available with AstLinux. Use the GUI to edit and mage the system, as AstLinux has a somewhat different directory structure than you may be familiar with You should be up and running in a couple of hours, have a low power 20 watts, fanless flash based system that will just work in a real case. The Pi is OK for a playtoy and some testing, but I much prefer the HP thin clients for a robust installation. I assume you are not doing any fancy call center or heavy database work. For a home or home office it is a really good solution. AstLinux is also used with other embedded installations on computers with multiple Ethernet ports, acting as router and firewall in addition. I prefer the component solution personally, which makes the HP thin clients the way to go. John Novack I have built more than 30 of these systems on various HP Thin Clients, used off of eBay with no failures Ira wrote: Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put it on a Raspberry Pi or something like that, but I need the one POTS line and all I have for that at the moment is a Digium card for a PCI slot. Are there any current thoughts on this? -- Ira -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Asterisk build
If you are really wanting to build something on Raspberry Pi or similar ARM platform, you could also take a look at Elastix for ARM. http://www.elastix.com/en/downloads/ Elastix is a fully integrated platform, and includes the majority of necessary components in one installation. The new Raspberry Pi 2 platform may be perfect for your needs in this respect, although based on your load, the B+ board may be more available at this time, and slightly cheaper. The Pi 2 is about double the core processing speed. YMMV Thanks, *Glenn Geller* *VDOTel* On Fri, Mar 6, 2015 at 12:34 PM, John Novack SCII jnov...@stromberg-carlson.org wrote: Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your older version of Asterisk to the fairly current version 11 currently available with AstLinux. Use the GUI to edit and mage the system, as AstLinux has a somewhat different directory structure than you may be familiar with You should be up and running in a couple of hours, have a low power 20 watts, fanless flash based system that will just work in a real case. The Pi is OK for a playtoy and some testing, but I much prefer the HP thin clients for a robust installation. I assume you are not doing any fancy call center or heavy database work. For a home or home office it is a really good solution. AstLinux is also used with other embedded installations on computers with multiple Ethernet ports, acting as router and firewall in addition. I prefer the component solution personally, which makes the HP thin clients the way to go. John Novack I have built more than 30 of these systems on various HP Thin Clients, used off of eBay with no failures Ira wrote: Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put it on a Raspberry Pi or something like that, but I need the one POTS line and all I have for that at the moment is a Digium card for a PCI slot. Are there any current thoughts on this? -- Ira -- Dog is my Co-Pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_pjsip ACL relation to endpoint
Hello. I continue to transfer chan_sip to pjsip. Friend in chan_sip can has options: deny=0.0.0.0/0.0.0.0 permit=192.168.0.1 pjsip offer to use global ACL without relation to any andpoint. My task is restriction via IP to registering in certain endpoint. Different rules to different endpoints. It will be better ACL has optional link to Endpoint. Or you can offer other solution? Dmitriy Serov -- _ -- 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] New Asterisk build
Hello John, Friday, March 6, 2015, 12:34:42 PM, you wrote: Find a HPT5720 with expansion chassis on eBay for under $50, load AstLinux ( instructions at AstLinux.org ) Move your Digium card and your confs , fix up any differences from your But given that means buying an old computer, why change at all? I already have a very low power one that works fine. Is AstLinux better than Centos 5 running Asterisk 13? -- Ira -- _ -- 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] res_pjsip endpoint config object's 'identify_by' option needs new value uri.
On Fri, Mar 6, 2015 at 3:46 PM, Dmitriy Serov serov@gmail.com wrote: 07.03.2015 0:24, Kevin Harwell пишет: On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov serov@gmail.com wrote: Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have match_auth_username=yes and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.comclient_uri=sip:1234567...@sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option contact_user. Incoming call from this registration will be INVITE sip:siptrunk-in@ I offer to change res_pjsip_endpoint_identifier_user to realize endpoint identification by sip uri. I think it will be usefull. P.S. i hope issues will be rejected: https://issues.asterisk.org/jira/browse/ASTERISK-22306 and SWP-6069 Dmitriy Serov - I believe what you are looking for is already available. See the identify type (type=identify) section that is in the pjsip.conf file and the identify option for endpoints. These allow you to identify and endpoint by IP address. For more information see the pjsip.conf.sample file. Also take a look at configuring Asterisk for res_pjsip [1] specifically the part about configuring endpoint identification by IP address [2]. If you run into problems more information can also be found in the res_pjsip troubleshooting guide [3], specifically the section on identify by IP address [1] https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Configuration_res_pjsip_endpoint_identifier_ip [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+PJSIP+Troubleshooting+Guide Hope that helps, Thank you for answer. But... ones again: I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? Simple Example. I have two trunks with their own credentials (and did) to the same sip server: - for home - for bussiness [home-example.com-endpoint] [bussiness-example.com-endpoint] [home-example.com-registration] contact_user=home-example.com-endpoint [bussiness-example.com-registration] contact_user=bussiness-example.com-endpoint ;and ok... i wrote identify by IP section [example.com-identify] type=identify match=example.com endpoint= ??? It is very! important for me to know what trunk passes through the incoming call: home or bussiness. 1. Identify by IP. Do you have answer? 2. Identify by username. What? I can't make endpoints to all of my contacts. Ok. I can use contact_user in registraction and route incoming call by INVITE uri. Can i? Dmitriy Serov I don't think I fully understand the scenario, but if you have different named endpoints originating from the same address (or even different addresses) then these can be identified by the username portion of the sip uri. However, if you have endpoints for instance with the same name, but different addresses then these can be distinguished by using the identity type. Your scenario looks like the first option. The endpoint names are different, but the address is the same, so identification based on the username should be sufficient. However, if you have a mix of both types on your system, for instance multiple endpoints on two different systems (IP addresses) with the same names, then I am unsure how you would select the correct endpoint even while attempting to identify_by uri. -- Kevin Harwell 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
Re: [asterisk-users] AWS/EC2 server selection
Hi Jeff Are you aware of any challenges of hosting it on AWS? It will help me to work out alternate plan. Is there any recommendation? Should I split it to multiple instances and balance traffic across multiple small server instances? I can use Kamailio to balance traffic. I see many posts referring to AWS deployment. Please help me to choose AWS server instance. *Thanks Regards,* Amit Patkar On 3/7/2015 12:19 AM, Jeff LaCoursiere wrote: Why use Amazon? With that kind of load I would want dedicated servers. Call Rackspace or Softlayer. j On 03/06/2015 11:59 AM, Amit Patkar wrote: Hi I plan to host Asterisk instances on AWS/EC2 servers. Requirement is to run asterisk instance with transcoding (g.729 + g.711) and full recording. Number of concurrent calls expected are 500+. 2 instances will be configured for 100% redundancy. Heart beat will be used to determine active instance. How should I choose EC2 instance? How many vCPU, RAM should be selected? I am assuming that server with ssd is required as all 500+ calls needs to be recorded. Regards, Amit Patkar -- _ -- 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
