[asterisk-users] UPDATE instead of RE-INVITE
HI,
It is possible to disable/remove INVITE method in 200 OK responses?
I want to receive from another SIP/PBX the the media path redirection in a
UPDATE message rather than an INVITE, after calls are transfered.
My asterisk is version 11.
e.g:
SIP/2.0 200 OK
.
From: user sip:+2404985962@IP;tag=1685058321
To: user2 sip:user2@10.20.0.22:5060;tag=as30297c0c
..
Allow: ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE, UPDATE
(INVITE should not be on the list for this 200 OK responses)
Now, the other endpoint won't see INVITE on 'Allow' list and should send
the media redirection path in a UPDATE sip message, instead of a INVITE
(according to the documentation).
Any idea how to do it in asterisk 11?
('canreinvite=update' doesn't seem to work on this version)
Regards,
--
Efficiency is doing things right; effectiveness is doing the right things
(Peter Drucker)
Miguel Oyarzo
DevOps VoIP Engineer
Linux User: # 483188 - counter.li.org
http://au.linkedin.com/in/mikeaustralia
Melbourne, Australia
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Re: [asterisk-users] Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?
You have to load the module res_srtp (secure media) in Asterisk. module load res_srtp.so (this is a requirement to talk websocket) If you don't have it, must build it and install it. Cheers, -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 6/18/2013 3:02 AM, Joel Rosenfield wrote: I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017) and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome. I have seen other people post a similar issue, but I have not seen a solution. If someone with good knowledge of this issue were to respond with this is a known issue or no, and this should be reported to Mozilla, that would be very helpful for me as well. Here is the error I see in the Asterisk console after it successfully parses the SDP a lines: Rejecting secure audio stream without encryption details: audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 Trying to put 'SIP/2.0 488' onto WS socket destined for www.xxx.yyy.zzz:5060 No compatible codecs for this SIP call. Here is the sip.conf info. I have tried various permutations of the dtls and encryption parameters with no luck. I do have openssl and srtp built into Asterisk (that solved a different error dealing with the RTP engine). [webrtc-dtls] ; Add DTLS stuff for Mozilla Nightly (and eventually Firefox) type=user host=dynamic hassip=yes transport=ws,wss directmedia=no ; proxy the media icesupport=yes ; needed for webrtc avpf=yes; needed for webrtc context=default encryption=yes dtlsenable=yes dtlsverify=no dtlsrekey=60 dtlscafile=/opt/asterisk/keys/ca.crt dtlscertfile=/opt/asterisk/keys/asterisk.pem dtlssetup=actpass insecure=invite Here is the SDP offered by Nightly: v=0 o=Mozilla-SIPUA-24.0a1 25687 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:7194cbcc a=ice-pwd:e57c14491015e529b84c5a6baf6d7b67 a=fingerprint:sha-256 48:3E:0C:59:BA:EB:6C:F9:5D:65:BF:08:54:63:C3:EA:AF:A9:60:9D:39:47:A5:41:6B:E1:A8:EB:7C:06:BE:D4 m=audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 c=IN IP4 www.xxx.yyy.zzz a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.1.109 62583 typ host a=candidate:1 1 UDP 1692467199 www.xxx.yyy.zzz 62583 typ srflx raddr 192.168.1.109 rport 62583 a=candidate:5 1 UDP 2111766783 192.168.56.1 62584 typ host a=candidate:0 2 UDP 2111832318 192.168.1.109 62585 typ host a=candidate:1 2 UDP 1692467198 www.xxx.yyy.zzz 62585 typ srflx raddr 192.168.1.109 rport 62585 a=candidate:5 2 UDP 2111766782 192.168.56.1 62586 typ host Thanks, - Joel -- _ -- 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] sipgate outgoing calls
It looks like the challenge response after INVITE is not been accepted. Provide more detail. $ sip set debug peer sipgate -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 7:10 PM, gpxctawjc...@irational.org wrote: On Thu, 19 Sep 2013, David Duffett wrote: i am getting these errors in asterisk cli -- Executing [01179553708@default:1] Set(SIP/-015b, CALLERID(num)=xx) in new stack -- Executing [01179553708@default:2] Dial(SIP/-015b, SIP/01179553708@sipgate,30,trg) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 01179553708@sipgate [Sep 19 10:08:03] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to ' sip:xx...@sipgate.co.uk;tag=as055d9532' -- SIP/sipgate-015c is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) any further ideas ? many thanks I believe registration is in place, otherwise inbound calls would not work. Also, registration is not required for outbound calls to work. I would suggest cutting down your sip.conf profile to this minimal configuration: host=sipgate.co.uk username=xxx fromuser=xxx insecure=invite,port secret=xxx context=my-inbound-context type=peer If outbound calls still do not with this, I would suggest that there may be an issue in the general section of your sip.conf Assuming calls do work, you can then add any other configuration lines you feel are necessary - but remember, as with all Asterisk configuration files, less is more :-) On 18 Sep 2013 22:06, Administrator TOOTAI ad...@tootai.net wrote: Le 18/09/2013 15:29, gpxctawjc...@irational.org a écrit : Hello Hi i am trying to setup sipgate gateway i can get incoming calls fine, but when i dial in and then try to dial out i get this in asterisk command line -- Called 01179248615@sipgate [Sep 18 13:58:30] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to '01179553708 sip:sip...@sipgate.co.uk;tag=as30eb9dd1' -- SIP/sipgate-014d is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) here is my sip.conf file [general] port = 5060 bindaddr = 0.0.0.0 context=default qualify=no disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm allow=slinear srvlookup=yes videosupport=yes alwaysauthreject=yes register = SIP-ID:sip-passw...@sipgate.co.uk/SIP-ID [sipgate] type=peer secret=SIP_PASSWORD insecure=invite username=SIP-ID defaultuser=SIP-ID fromuser=SIP-ID context=sipgate_in fromdomain=sipgate.co.uk host=sipgate.co.uk outboundproxy=proxy.live.sipgate.co.uk qualify=yes disallow=all allow=alaw dtmfmode=rfc2833 SIP-ID:SIP-Password obviously, i replace these with my login details but, are these the same thing ? SIP-Password SIP_PASSWORD the sipgate guides are contradictory http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk http://www.live.sipgate.co.uk/faq/article/508/How_do_I_configure_Asteri sk any suggestions ? Many thanks My setup with sipgate.de [sipgate] type=peer secret=MY-PASSWORD defaultuser=SIP-ID host=217.10.79.9 fromuser=SIP-ID fromdomain=sipgate.de context=incoming-sipgate ;qualify=900 dtmfmode=info directmedia=yes insecure=port,invite disallow=all allow=ulaw,alaw accountcode=MY-ACCOUNTCODE What you forget is to register with them: ; Sipgate register = SIP-ID:my-passw...@sipgate.de/SIP-ID ;don't accept to register without FQDN Hope that help -- 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
Re: [asterisk-users] sipgate outgoing calls
Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 7:55 PM, gpxctawjc...@irational.org wrote: It looks like the challenge response after INVITE is not been accepted. Provide more detail. $ sip set debug peer sipgate server*CLI sip set debug peer sipgate SIP Debugging Enabled for IP: 217.10.79.23:5060 Really destroying SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' Method: REGISTER -- Registered SIP 'x' at 86.140.115.135 port 5060 == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Executing [01179553708@default:1] Set(SIP/x-015d, CALLERID(num)=x) in new stack -- Executing [01179553708@default:2] Dial(SIP/x-015d, SIP/01179553708@sipgate,30,trg) in new stack == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 -- Called 01179553708@sipgate [Sep 19 10:50:15] NOTICE[28232]: chan_sip.c:17885 handle_response_invite: Failed to authenticate on INVITE to 'x sip:xx...@sipgate.co.uk;tag=as629ee6f8' -- SIP/sipgate-015e is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [01179553708@default:3] Hangup(SIP/x-015d, ) in new stack == Spawn extension (default, 01179553708, 3) exited non-zero on 'SIP/x-015d' --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK73a1638c;rport=5060 From: asterisk sip:asterisk@92.63.131.3;tag=as5dcb32d8 To: sip:sipgate.co.uk;tag=99199803810c7e807ea44745826d9aa4.df2d Call-ID: 65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 - --- (11 headers 0 lines) --- Really destroying SIP dialog '65cb07675eefaaef5f655e8a0be6b2f6@92.63.131.3' Method: OPTIONS [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:11588 sip_reregister: -- Re-registration for xxx...@sipgate.co.uk REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 217.10.79.23:5060: REGISTER sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK13b202d7;rport Max-Forwards: 70 From: sip:x...@sipgate.co.uk;tag=as19513575 To: sip:xx...@sipgate.co.uk Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1 CSeq: 182 REGISTER User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Authorization: Digest username=xx, realm=sipgate.co.uk, algorithm=MD5, uri=sip:sipgate.co.uk, nonce=523ac9531b1cc7962e07bce6a76683ee24da44d0, response=c82fac231a41085c275899ad84f73317 Expires: 120 Contact: sip:xx@92.63.131.3 Content-Length: 0 --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;received=92.63.131.3;branch=z9hG4bK13b202d7;rport=5060 From: sip:xx...@sipgate.co.uk;tag=as19513575 To: sip:xx...@sipgate.co.uk;tag=c3e497ecaece77a8e244e564b4212178.3e46 Call-ID: 3ef8ff1a6ec360626af409b112b860ee@127.0.1.1 CSeq: 182 REGISTER Contact: sip:xx@92.63.131.3;expires=120 Content-Length: 0 - --- (8 headers 0 lines) --- Scheduling destruction of SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1' in 32000 ms (Method: REGISTER) [Sep 19 10:51:05] NOTICE[28232]: chan_sip.c:18301 handle_response_register: Outbound Registration: Expiry for sipgate.co.uk is 120 sec (Scheduling reregistration in 105 s) Reliably Transmitting (no NAT) to 217.10.79.23:5060: OPTIONS sip:sipgate.co.uk SIP/2.0 Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport Max-Forwards: 70 From: asterisk sip:asterisk@92.63.131.3;tag=as5afd24b2 To: sip:sipgate.co.uk Contact: sip:asterisk@92.63.131.3 Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Date: Thu, 19 Sep 2013 09:51:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- server*CLI --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 92.63.131.3:5060;branch=z9hG4bK2d73294a;rport=5060 From: asterisk sip:asterisk@92.63.131.3;tag=as5afd24b2 To: sip:sipgate.co.uk;tag=99199803810c7e807ea44745826d9aa4.c753 Call-ID: 1becd4dc336869c4692fc4e55e109562@92.63.131.3 CSeq: 102 OPTIONS Accept: */* Accept-Encoding: Accept-Language: en Supported: Content-Length: 0 - --- (11 headers 0 lines) --- Really destroying SIP dialog '1becd4dc336869c4692fc4e55e109562@92.63.131.3' Method: OPTIONS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs
Re: [asterisk-users] sipgate outgoing calls
What you don't have mentioned yet is whether your outbound call reaches the destination. -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/19/2013 8:26 PM, gpxctawjc...@irational.org wrote: On Thu, 19 Sep 2013, Miguel Oyarzo wrote: Challenge authentication look good. --- SIP read from UDP:217.10.79.23:5060 --- SIP/2.0 200 OK Are you sure this number format 01179553708 is accepted in that SIP trunk? Some VOIP providers only accept international format. when i use a softphone client to connect directly to sipgate i can dial 01179553708 and get through -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.8 sends SIP/2.0 481 Call/Transaction Does Not Exist to INVITE
To: sip:8009499...@x.yyy.32.10:5060;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 In your call sample To has a tag. if this is the first Invite it can't have a tag at the end, otherwise Asterisk will look for an existing dialog in its database and will show an error, if can't find any. It looks like the other end is never closing the previous dialog?.. is Asterisk sending a proper 200 OK after receiving a BYE? NAT problem? regards, -- == Miguel Oyarzo DevOps Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 9/17/2013 6:18 AM, Vik Killa wrote: Asterisk is sending a 481 in response to an INVITE for reasons I do not understand. Here is the INVITE: INVITE sip:8009499...@x.yyy.32.3:5060;transport=udp SIP/2.0 Record-Route: sip:X.YYY.32.10;lr=on;ftag=247898 To: sip:8009499...@x.yyy.32.10:5060;tag=ac86f72d2bfe10395b2e62e01c70bf66.0f65 From: Scott Thompson sip:7166359...@x.yyy.32.10;tag=247898 Via: SIP/2.0/UDP X.YYY.32.10;branch=z9hG4bK542e.5042d534.0 Via: SIP/2.0/UDP X.YYY.33.178:5060;rport=5060;received=X.YYY.33.178;branch=z9hG4bK57b720cccb00f8498662f48e8 Call-ID: 94f80f866e877490729548a079abe371@192.168.101.5 mailto:94f80f866e877490729548a079abe371@192.168.101.5 CSeq: 2 INVITE Contact: sip:7166359...@x.yyy.33.178:5060 Max-Forwards: 69 x-inin-crn: 2001471530;loc=Amherst;ms=STAMPEDE-MS Supported: join, replaces User-Agent: ININ-TsServer/3.13.11.12748 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, SUBSCRIBE Accept: application/sdp Accept-Encoding: identity Content-Type: application/sdp Content-Length: 252 Proxy-Authorization: Digest username=909003660716,realm=X.YYY.32.10,nonce=523755911a22ed0fae66765d46ef9131e311fbb9d2fb,uri=sip:8009499...@x.yyy.32.10:5060,response=cb6306569b3047ac35064dcb5aee6db4 X-Enswitch-RURI: sip:8009499...@x.yyy.32.10:5060 X-Enswitch-Source: X.YYY.33.178:5060 The only problem I see with this INVITE is the VIAs are not right after the INVITE line... although in https://www.ietf.org/rfc/rfc3261.txt, it explicitly states the the order of the headers is not a requirement, it seems Asterisk does make it one... The relative order of header fields with different field names is not significant. However, it is RECOMMENDED that header fields which are needed for proxy processing (Via, Route, Record-Route, Proxy-Require, Max-Forwards, and Proxy-Authorization, for example) appear towards the top of the message to facilitate rapid parsing. The relative order of header field rows with the same field name is important. -- _ -- 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] Sending SMS from asterisk
chan_datacard was discontinnued two years ago, chan_dongle is the current dongle driver for asterisk. chan_mobile uses bluetooth mobile phones as FXO devices. to send SMS chan_dongle should be used. i.e: asterisk -rx [ENTER] dongle sms dongle0 0415340999 hello world this command will send and SMS to 0415340999 by dongle0. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/14/2013 9:15 AM, Asghar Mohammad wrote: HI bilal, I don't think DAHDI can send SMS you have 2 options chan_mobile or chan_datacard ex chan_dongle chan_datacard i have not tested but with some mobile phones you can send sms i have tested also with some made in china unbranded phone that are capable to send and receive sms but not good for call termination, they send answer on connect. not all BT dongles are compatible you should go to trail and error for finding combination of dongle and phone. PS: yesterday tested asterisk 11 with chan_mobile and worked without any modification. On Wed, Mar 13, 2013 at 10:29 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi Asghar; I was looking to use chan_mobile for sending SMS, is it possible? Or it is only for calls? By the way, if I have GSM adaptor that convert from SIM card to FXS port, then who I need chan_mobile? I can use DAHDI. So when to use chan_mobile? Regards Bilal - HI Bilal, i am using chan_mobile for call termination, you can use it but you need to tweak chan_mobile.c it is broken from a long time. let me know if you want give it a try. On Mon, Mar 11, 2013 at 6:22 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: - What are the elements of this solution? Is it only: 3G dongles and chan_dongle only? Or there are something else? Bash and perl programing, asterisk and chan_dongle. * Bash and perl programing to do what? It is going to use AMI instead of sending the messages from the commands given in the extensions.conf? Why to use chan_dongle and not chan_mobile? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Sending SMS from asterisk
What are the elements of this solution? Is it only: 3G dongles and chan_dongle only? Or there are something else? Bash and perl programing, asterisk and chan_dongle. About the script that you wrot it: This script is using asterisk (through AMI) to send the SMS? Or it is working without need for asterisk? In this case, where is the benifit of using Asterisk to send the SMS? It uses the asterisk API. The only benefit I found was the queue management for high volume of sending/receiving. As asterisk can't send SMS more than160-characters, I had to work out how to do it (PDU alteration). Anyway, 160-characters limit should be enough for most users. Cheers! -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/10/2013 2:10 PM, bilal ghayyad wrote: Dears; We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. What are the elements of this solution? Is it only: 3G dongles and chan_dongle only? Or there are something else? About the script that you wrot it: This script is using asterisk (through AMI) to send the SMS? Or it is working without need for asterisk? In this case, where is the benifit of using Asterisk to send the SMS? Regarding to Sangoma and khomp: Do u mean that they have something like Huwewi 3G dongles? Regards Bilal -- Hi Bilal, It's not necessary to use a FXS port, you can compile install chan_dongle and buy a Huawei 3G dongle. We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. In addition, I wrote an script able to send up to 12000 characters in concatenated SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM, Gerardo Barajas wrote: Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com http://www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Sending SMS from asterisk
Hi Bilal, It's not necessary to use a FXS port, you can compile install chan_dongle and buy a Huawei 3G dongle. We have running here a SMS solution with four 3G dongles, which sends over 20.000 SMS a month. In addition, I wrote an script able to send up to 12000 characters in concatenated SMS (the recipient receives a single SMS only) chan_dongle works very well. -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 3/9/2013 1:09 PM, Gerardo Barajas wrote: Yes, you can check solutions from sangoma and khomp. Saludos/Regards -- Ing. Gerardo Barajas Puente Proyectos Especiales/Preventa | www.neocenter.com http://www.neocenter.com T:+52 (55) 8590-9000 x 7003 On Fri, Mar 8, 2013 at 6:32 PM, bilal ghayyad bilmar...@yahoo.com mailto:bilmar...@yahoo.com wrote: Hi; If my landline service provider does not provide the ability to send the SMS, and I need to send SMS from asterisk, then what is the required? How? Is it possible to send SMS from asterisk using SIM card to be connected via GSM adaptor connected to FXS ports? Or HOW? From the other side, this is existed only in asterisk 1.8 or it is existed in asterisk 1.4? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 firmware for centos 6
Did you see this URL? http://downloads.asterisk.org/pub/telephony/asterisk/ On Wed, Nov 14, 2012 at 4:24 AM, Justin Killen jkil...@allamericanasphalt.com wrote: In http://packages.digium.com/centos/ there is not yet a centos 6 branch (Nor is there a RHEL 6 branch). Centos 6.0 was release in July of 2011 – is this something that Digium is planning on supporting? Or is there a different URL that I’m not aware of for firmware packages? ** ** -Justin Killen -- _ -- 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 -- ==** Miguel Oyarzo Senior [Network Systems Design] Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia ==** -- _ -- 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 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
It seems a firewall or signaling problem. The calling part is not sending a ACK response to your host because it never get an OK 200 from your host. In other words, the called part is trying to send to the calling part a) TRYING 100, then b) RING 180 and finally c) OK 200 but the calling part seems not being receiving no signals from you host. As a result, your host has sent 4 times SIP/2.0 200 OK (retransmissions) to the calling part but it never got an ACK from the other end to establish the communication. Then, the link is destroyed. regards, -- == Miguel Oyarzo Senior [ Network | Systems Design ] Engineer http://www.linkedin.com/in/mikeaustralia Linux User: # 483188 - counter.li.org Melbourne, Australia On 11/11/2012 9:46 PM, Eric Kuhnke wrote: Hi all, I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: One Yealink SIP-T22P phone (10.0.0.107), extension 10, configured to talk to my local FreePBX/Asterisk 11.0 server which is at 10.0.0.17. The Yealink phone doesn't seem to have any problem placing outgoing calls through the Asterisk server, which is registered to Diamondcard. I can reach both the Asterisk server itself (for example to use voicemail) or call any number on the PSTN. Likewise I have the server configured to pass incoming DID calls for myDIDnumber to extension 10. Calls from the PSTN to myDIDnumber ring the phone, including CID passing, and will connect a full duplex audio call session. The problem is that the phone won't stay connected longer than 13 to 17 seconds. When the phone is manually configured to use my account and password on the diamondcard servers directly, both incoming and outgoing calls work normally, with RTP/UDP port 5060 traffic passing through my NAT without trouble. I have made no special modifications to the NAT. 13 seconds after picking up an incoming call, the phone disconnects at the same time as the log shows this: [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Retransmission timeout reached on transmission08a728706baea3b740aa806e41e9d13d@69.71.222.196 mailto:08a728706baea3b740aa806e41e9d13d@69.71.222.196 for seqno 103 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 17853ms with no response [2012-11-11 01:51:40] WARNING[3201] chan_sip.c: Hanging up call 08a728706baea3b740aa806e41e9d13d@69.71.222.196 mailto:08a728706baea3b740aa806e41e9d13d@69.71.222.196 - no reply to our critical packet (seehttps://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). *The full log and configuration is at:* *http://pastebin.com/1Mgn72vN* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11 / FreePBX: Incoming calls timeout after 13 seconds, outgoing works perfectly
On Mon, Nov 12, 2012 at 11:17 AM, Markus unive...@truemetal.org wrote: Am 11.11.2012 11:46, schrieb Eric Kuhnke: I'm trying to troubleshoot an issue with my SIP service. All outgoing calls work normally. The following is a SIP debug log from Asterisk. The test setup is as follows: Miguel already explained what's going on. Have a look at the SIP packets to figure out more. On the Asterisk box: tcpdump -nnqt -s 0 -A -i eth0 port 5060 Also, check your router/firewall logs, respectively activate them, to find out why the packets are not going through. Maybe also try qualify=yes yes, correct. In addition, He might being notifying to the calling part incorrectly about the called part is behind a nat :) http://pastebin.com/1Mgn72vN (line 71: nat=no) -- ==** Miguel Oyarzo Senior Systems Design Engineer Linux User: # 483188 - counter.li.org http://au.linkedin.com/in/mikeaustralia Melbourne, Australia ==** -- _ -- 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
