Re: [asterisk-users] Asterisk as TLS server as well as TLS client
Le 20/08/2012 17:02, Daniel Pocock a écrit : On 20/08/12 16:23, Administrator TOOTAI wrote: Hi, I have to connect 3 asterisk servers,each of them being TLS server for his clients and connected in both way in TLS with both others asterisk, each having hi own Common Name. Is this possible? I set up 2 asterik's , one server and the other client, this is OK. But I can't deal with certificats generated on both servers. I tried to put tlscertfile ans tlscafile in the peer definition, each pointing to the certificate generated by the server, but thatś not working. Thanks for any hint. Asterisk doesn't seem to implement mutual TLS authentication, see the comments in this thread: http://java.net/projects/jitsi/lists/users/archive/2012-08/message/37 People who want strong TLS typically use a SIP proxy as a front-end to Asterisk, either repro or Kamailio stand out as leaders in TLS support http://www.opentelecoms.org/use-a-sip-proxy-instead-of-asterisk At the bottom, there are links to some practical guides how to use either repro or Kamailio with Asterisk Thanks for those informations. Regards -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] alwaysauthreject=yes not working as expected
Asterisk 1.4.42 Set alwaysauthreject=yes in [general] section of sip.conf. Restarted asterisk However when I attempt to register I still get: [2012-08-08 21:11:34] NOTICE[15689] chan_sip.c: Registration from 'sip:000333082261...@domain.com' failed for '121.98.1.1' - Wrong password [2012-08-08 21:12:42] NOTICE[15689] chan_sip.c: Registration from 'sip:00033308226...@domain.com' failed for '121.98.1.1' - No matching peer found Based on the Asterisk security advisory (http://downloads.asterisk.org/pub/security/AST-2011-011.html) I would have expected 1.4.42 to respond the same in both cases (since the issue was fixed in 1.4.41.2). Am I missing something obvious? Yes. Those are log messages for the administrator's benefit. They are not SIP messages sent in response to the REGISTER request. The SIP messages sent are supposed to be the same not the logging messages. Yes I agree they are supposed to be the same but they are not. Below is the dialog when a wrong password is provided with alwaysauthreject=yes: U 121.98.1.1:1025 - 203.89.1.1:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;rport..Max-Forwards: 70..C ontact: sip:1232261336@192.168.1.103:5060;rinstance=da68419a02006162. .To: sip:1232261...@domain.com..From: sip:123 2261...@domain.com;tag=f910aa53..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANC EL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 5.0.0 stamp 67284..Content-Length: 0 U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip:000333 082261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY 2E...CSeq: 1 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip: 1232261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com;tag=as16fea110..Call- ID: ZmM4YTU4NTg2MWNhYzVk YTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 1 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: repla ces..WWW-Authenticate: Digest algorithm=MD5, realm=domain.com, nonce=2f48b121..Content-Length: 0 U 121.98.1.1:1025 - 203.89.1.1:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;rport..Max-Forwards: 70..C ontact: sip:1232261336@192.168.1.103:5060;rinstance=da68419a02006162. .To: sip:1232261...@domain.com..From: sip:123 2261...@domain.com;tag=f910aa53..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 2 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANC EL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 5.0.0 stamp 67284..Authorization: Digest username=1232261336,re alm=domain.com,nonce=2f48b121,uri=sip:c-vm- 02.domain.com,response=cb74a7805412a3ac198800aeede3c06e,algorit hm=MD5..Content-Length: 0 U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip:000333 082261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY 2E...CSeq: 2 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Length: 0 SIP/2.0 403 Forbidden (Bad auth)..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;received=121.98.1.1;rport=1025..Fro m: sip:1232261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com;tag=as16fea110..Call-ID: ZmM4YTU4NTg2 MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 2 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supporte d: replaces..Content-Length: 0 Is this a bug or am I missing something obvious? -- _ -- 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] version compatible with centos 5.7 (2.6.18-308.8.2.el5)
dear guys plz tell me which version of asterisk is compatible with centos 5.7 (2.6.18-308.8.2.el5). and which is the latest version. regards neo -- _ -- 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] version compatible with centos 5.7 (2.6.18-308.8.2.el5)
On 12-08-21 07:04 AM, neo nortan wrote: dear guys plz tell me which version of asterisk is compatible with centos 5.7 (2.6.18-308.8.2.el5). and which is the latest version. Almost any version of asterisk should compile on any recent version of CentOS. You can find the latest versions of Asterisk at http://www.asterisk.org/downloads -- Looking for (employment|contract) work in the Internet industry, preferrably working remotely. Building / Supporting the net since 2400 baud was the hot thing. Ask for a resume! ispbuil...@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] Which card to get?
We are investigating the possibility of using Asterisk in a KVM based virtual machine to handle connections to and from our HylaFax service. Our current set up uses a dedicated host with external fax modems. What I wish to know is what interface card would the list members recommend for a proof of concept trial? We currently have two incoming fax lines and five vox lines all POTS. Our physical internet connection is fiber but I could not tell you exactly what type of service it presently carries. It is upgradable to a considerable extent in any case. We are planning to move to VOIP as an adjunct to this project. This is secondary to getting the fax system moved but we would like to avoid having to install additional hardware for VOIP once the fax portion of project is complete and the service transferred. What are our options? -- *** E-Mail is NOT a SECURE channel *** James B. Byrnemailto:byrn...@harte-lyne.ca Harte Lyne Limited http://www.harte-lyne.ca 9 Brockley Drive vox: +1 905 561 1241 Hamilton, Ontario fax: +1 905 561 0757 Canada L8E 3C3 -- _ -- 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] Which card to get?
We are investigating the possibility of using Asterisk in a KVM based virtual machine We have found that Asterisk in a VM with a digital card does not work (At least under VMWare ESXi 5 with PCI pass though). The timing on the card tested with dahdi_test were awful, sub 80% on average. The same card on a bare metal machine was %99.998 on average. 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
Re: [asterisk-users] Which card to get?
On Tue, Aug 21, 2012 at 09:33:39AM -0400, James B. Byrne wrote: We are investigating the possibility of using Asterisk in a KVM based virtual machine to handle connections to and from our HylaFax service. Our current set up uses a dedicated host with external fax modems. What I wish to know is what interface card would the list members recommend for a proof of concept trial? Hopefully if someone else has a different experience they will speak up, but I've not personally heard of any cards working reliably in a PCI passthrough mode. This could e a problem if you intende the KVM guest to handle all the card communication. You could get an analog gateway, or setup another small system to host any cards you get to act as a gateway, and keep most of your logic on the KVM. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which card to get?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Tuesday, August 21, 2012 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Which card to get? On Tue, Aug 21, 2012 at 09:33:39AM -0400, James B. Byrne wrote: We are investigating the possibility of using Asterisk in a KVM based virtual machine to handle connections to and from our HylaFax service. Our current set up uses a dedicated host with external fax modems. What I wish to know is what interface card would the list members recommend for a proof of concept trial? Hopefully if someone else has a different experience they will speak up, but I've not personally heard of any cards working reliably in a PCI passthrough mode. This could e a problem if you intende the KVM guest to handle all the card communication. You could get an analog gateway, or setup another small system to host any cards you get to act as a gateway, and keep most of your logic on the KVM. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org This is not the route I would necessarily recommend, but the OBI110 works for us as a POTS/DAHDI gateway on our VM setup. You use 1 OBI110 box for each pots line so you would need about 6 as I understand your post. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test Websocket support in SIP in Asterisk trunk?
On Mon, Aug 20, 2012 at 8:20 PM, mailsvb mail...@gmail.com wrote: Hi, you need to build Asterisk with SRTP support... wget http://sourceforge.net/projects/srtp/files/latest/download -O srtp-latest.tgz tar -zxvf srtp-latest.tgz ./configure --prefix=/libsrtp make make install And for Asterisk... ./configure --with-srtp=/libsrtp this should work... Recompiled. Well... now at leat in ONE instance the signaling seems to behave correctly: when I dial from sipml5 to plain SIP. If the destination is sipml5, the destination browser goes into a funky state in which the live camera panel pops up but there doesn't seem to be a recognized ringing state. Here's the log from a sipml5-sipml5 call. The caller is 2010 and the callee is 2009. (12:40:06 is when I gave up and clicked hangup at the caller.) (Media? Heh, surely you jest.) [Aug 21 12:38:25] DEBUG[22872] res_timing_timerfd.c: Expected to acknowledge 1 ticks but got 6 instead [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: = Looking for Call ID: e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 (Checking From) --From tag wiwN3MEMrB3HGUmlel5V --To-tag [Aug 21 12:38:35] DEBUG[23469] logger.c: CALL_ID [C-0002] created by thread. [Aug 21 12:38:35] DEBUG[23469] acl.c: For destination '192.168.0.92', our source address is '192.168.0.111'. [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: Setting SIP_TRANSPORT_WS with address 192.168.0.111:5060 [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: Allocating new SIP dialog for e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 - INVITE (No RTP) [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: CALL_ID [C-0002] bound to thread. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Trying to put 'SIP/2.0 401' onto WS socket destined for 192.168.0.92:5060 [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: Call_ID [C-0002] being removed from thread. [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: = Looking for Call ID: e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 (Checking From) --From tag wiwN3MEMrB3HGUmlel5V --To-tag as39a7b995 [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: CALL_ID [C-0002] bound to thread. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Received ACK (6) - Command in SIP ACK [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Stopping retransmission on 'e4d7cda4-c4cb-932f-c084-ac6f87d27eb9' of Response 3106: Match Not Found [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: Call_ID [C-0002] being removed from thread. [Aug 21 12:38:35] DEBUG[23469] chan_sip.c: = Looking for Call ID: e4d7cda4-c4cb-932f-c084-ac6f87d27eb9 (Checking From) --From tag wiwN3MEMrB3HGUmlel5V --To-tag [Aug 21 12:38:35] DEBUG[23469] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] logger.c: CALL_ID [C-0002] bound to thread. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Received INVITE (5) - Command in SIP INVITE [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0xb751d8dc' [Aug 21 12:38:35] DEBUG[23469][C-0002] res_rtp_asterisk.c: Allocated port 18704 for RTP instance '0xb751d8dc' [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] DEBUG[23469][C-0002] rtp_engine.c: RTP instance '0xb751d8dc' is setup and ready to go [Aug 21 12:38:35] DEBUG[23469][C-0002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0xb751d8dc' [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: Splitting '192.168.0.111' into... [Aug 21 12:38:35] DEBUG[23469][C-0002] netsock2.c: ...host '192.168.0.111' and port ''. [Aug 21 12:38:35] VERBOSE[23469][C-0002] netsock2.c: == Using SIP RTP CoS mark 5 [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Setting NAT on RTP to Off [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Processing session-level SDP o=- 1190078527 1 IN IP4 127.0.0.1... UNSUPPORTED OR FAILED. [Aug 21 12:38:35] DEBUG[23469][C-0002] chan_sip.c: Processing session-level SDP s=webrtc (chrome 22.0.1189.0) - Doubango Telecom (sipML5 r000)... UNSUPPORTED OR FAILED.
Re: [asterisk-users] alwaysauthreject=yes not working as expected
- Original Message - From: CB kj...@xnet.co.nz To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, August 21, 2012 4:39:32 AM Subject: Re: [asterisk-users] alwaysauthreject=yes not working as expected Asterisk 1.4.42 First, even if you were right and you discovered a security vulnerability in Asterisk 1.4.42, that version of Asterisk is now in EOL, and no new security releases will be made. https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions You would of course be more then welcome to solicit patches from the open source community, but no new version of Asterisk 1.4.x would be released. snip Yes I agree they are supposed to be the same but they are not. Below is the dialog when a wrong password is provided with alwaysauthreject=yes: U 121.98.1.1:1025 - 203.89.1.1:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;rport..Max-Forwards: 70..C ontact: sip:1232261336@192.168.1.103:5060;rinstance=da68419a02006162. .To: sip:1232261...@domain.com..From: sip:123 2261...@domain.com;tag=f910aa53..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 1 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANC EL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 5.0.0 stamp 67284..Content-Length: 0 snip U 203.89.1.1:5060 - 121.98.1.1:1025 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-d88996fba8b1fd8c-1---d8754z- ;received=121.98.1.1;rport=1025..From: sip: 1232261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com;tag=as16fea110..Call- ID: ZmM4YTU4NTg2MWNhYzVk YTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 1 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: repla ces..WWW-Authenticate: Digest algorithm=MD5, realm=domain.com, nonce=2f48b121..Content-Length: 0 This is expected behavior. U 121.98.1.1:1025 - 203.89.1.1:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;rport..Max-Forwards: 70..C ontact: sip:1232261336@192.168.1.103:5060;rinstance=da68419a02006162. .To: sip:1232261...@domain.com..From: sip:123 2261...@domain.com;tag=f910aa53..Call-ID: ZmM4YTU4NTg2MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 2 REGISTER..Expires: 3600..Allow: INVITE, ACK, CANC EL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..User-Agent: X-Lite release 5.0.0 stamp 67284..Authorization: Digest username=1232261336,re alm=domain.com,nonce=2f48b121,uri=sip:c-vm- 02.domain.com,response=cb74a7805412a3ac198800aeede3c06e,algorit hm=MD5..Content-Length: 0 snip SIP/2.0 403 Forbidden (Bad auth)..Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-d8754z-5c88940128ede618-1---d8754z- ;received=121.98.1.1;rport=1025..Fro m: sip:1232261...@domain.com;tag=f910aa53..To: sip:1232261...@domain.com;tag=as16fea110..Call-ID: ZmM4YTU4NTg2 MWNhYzVkYTBhN2Q2MjA1YmUyMmYzY2E...CSeq: 2 REGISTER..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supporte d: replaces..Content-Length: 0 Is this a bug or am I missing something obvious? That is expected behavior as well. ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, ; for any reason, always reject with an identical response ; equivalent to valid username and invalid password/hash ; instead of letting the requester know whether there was ; a matching user or peer for their request. This reduces ; the ability of an attacker to scan for valid SIP usernames. ; This option is set to yes by default. The 401 response merely indicates that some level of authorization is required. The 403 response matches what would be sent if the username was valid but an invalid password/hash was provided. This response should be sent regardless if the username was actually valid. Based on your provided SIP traffic, that appears to be what happened. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Check for the voicemail
Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do? :( Danilo -- _ -- 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] Check for the voicemail
Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Tuesday, August 21, 2012 11:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Check for the voicemail Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do? :( Danilo -- _ -- 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] Check for the voicemail
just another thought: if you send the message by mail, do you need to save it? regards, Ruben Am 21.08.2012 18:45, schrieb Danny Nicholas: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danilo Dionisi Sent: Tuesday, August 21, 2012 11:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Check for the voicemail Hi all, I have a problem with voicemail. My boss has asked me to send via email, the message that a user leaves on the voicemail. This is very easy. :) After, he asked me to check before sending the email, if the receiver's mailbox is full. If the mailbox is full, Asterisk should call the receveir intern (example 2001) and using a Playback tell him that his mailbox is full. How can I do? :( Danilo -- _ -- 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] Check for the voicemail
I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- -- _ -- 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] Check for the voicemail
The message must be deleted if sent to the recipient, otherwise it must remain on the Asterisk machine when the recipient's mailbox is full. Il 21/08/12 18:52, Ruben Rögels ha scritto: just another thought: if you send the message by mail, do you need to save it? regards, Ruben -- -- _ -- 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] Check for the voicemail
Hello Check voicemail.conf maxmsg = 100 And change it. On Tue, Aug 21, 2012 at 12:52 PM, Danilo Dionisi dionisi.dan...@gmail.com wrote: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- -- _ -- 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] Check for the voicemail
I'll explain. I have an email account, danilo.dionisi @ outlook.it, with a maximum size of 100MB. For example, my inbox is full, and Paris Hilton ( =P ) leaves me a voicemail message. I have to check the space of my inbox, this space is completely full, so I do not have to delete the voicemail message but I have to call my SIPphone and via a Playback announce that my inbox is full. Il 21/08/12 18:55, Carlos Rojas ha scritto: Hello Check voicemail.conf maxmsg = 100 And change it. On Tue, Aug 21, 2012 at 12:52 PM, Danilo Dionisi dionisi.dan...@gmail.com wrote: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- -- _ -- 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] Check for the voicemail
Okay, so have a look at mailcmd= option in voicemail.conf mailbox will mean a e-mail-box in the next lines. What you need to do is wirting a shell script or what ever to check for the return code of the smtp session (normally it should be a 450 in case of full mailbox). In case of 450 mailbox full whatsoever you need to create a call file in /var/spool/asterisk/outgoing directing your recipient phone to a special extension which handles the playback of the your mailbox is full message. But: this will only work if you deliver the mail directly to the recipients mail server, if you use a smart host, it will accept the message and the final recipient server will bounce the mail back to you. I hope this helps. regards, Ruben Am 21.08.2012 18:52, schrieb Danilo Dionisi: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- _ -- 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] Check for the voicemail
I'm sorry, how i can to check for the return code of the smtp session? I've never done :p Thanks, Danilo Il 21/08/12 19:05, Ruben Rögels ha scritto: Okay, so have a look at mailcmd= option in voicemail.conf mailbox will mean a e-mail-box in the next lines. What you need to do is wirting a shell script or what ever to check for the return code of the smtp session (normally it should be a 450 in case of full mailbox). In case of 450 mailbox full whatsoever you need to create a call file in /var/spool/asterisk/outgoing directing your recipient phone to a special extension which handles the playback of the your mailbox is full message. But: this will only work if you deliver the mail directly to the recipients mail server, if you use a smart host, it will accept the message and the final recipient server will bounce the mail back to you. I hope this helps. regards, Ruben Am 21.08.2012 18:52, schrieb Danilo Dionisi: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- _ -- 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] confbridge
On 08/17/2012 03:21 PM, Jerry Geis wrote: On 08/17/2012 06:36 AM, Jerry Geis wrote: On 08/13/2012 04:58 PM, Jerry Geis wrote: On 08/13/2012 01:13 PM, Jerry Geis wrote: I am getting a beep beep beep (like a busy or hangup sound) when I am using my AGI to start up a conf. (did not happen with Meetme). The confbridge works, but the beep beep beep is mixed in with the audio. I have turned off every sound in the confbridge.conf file. How can I find out where this beep, beep beep is coming from and turn it off??? Jerry Jonathan and Danny I am only using Local channels and SIP to another asterisk box - one at this time. No phones no anything. I think is happening when the AGI is exiting. Its does not happen when I manually start it. --- here is the CLI [Kdevgeis*CLI [0K-- Executing [app_confbridge_call_out@smvoice-local-public-address-playfile:1] Set(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;2, agi_use_meetme=0) in new stack [Kdevgeis*CLI [0K-- Executing [app_confbridge_call_out@smvoice-local-public-address-playfile:2] AGI(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;2, smvoice,-digium_success,-pa_list) in new stack [Kdevgeis*CLI [0K-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Kdevgeis*CLI [0K == Manager 'MessageNet' logged on from 127.0.0.1 [Kdevgeis*CLI [0K == Manager 'MessageNet' logged off from 127.0.0.1 [Kdevgeis*CLI [0K == Manager 'MessageNet' logged off from 127.0.0.1 [Kdevgeis*CLI [0K == Manager 'MessageNet' logged on from 127.0.0.1 [Kdevgeis*CLI [0K[Aug 13 16:53:57] ERROR[17501]: utils.c:1221 ast_careful_fwrite: fwrite() returned error: Broken pipe == Manager 'MessageNet' logged off from 127.0.0.1 [Kdevgeis*CLI [0K == Using SIP RTP CoS mark 5 [Kdevgeis*CLI [0K --Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;2AGI Script smvoice completed, returning 0 [Kdevgeis*CLI [0K-- Executing [app_confbridge_call_out@smvoice-local-public-address-playfile:3] Wait(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;2, 1) in new stack [Kdevgeis*CLI [0K Channel SIP/devgeis_to_ebox4300-0003 was answered. [Kdevgeis*CLI [0K-- Executing [smvoice_pa_app_confbridge_intercom@smvoice-transfers:1] ConfBridge(SIP/devgeis_to_ebox4300-0003, PA0001,MessageNetConfBridge,MessageNetConfUser) in new stack [Kdevgeis*CLI [0K-- Executing [app_confbridge_call_out@smvoice-local-public-address-playfile:4] NoOp(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;2, list=PA0001) in new stack [Kdevgeis*CLI [0K-- Executing [app_confbridge_call_out@smvoice-local-public-address-playfile:5] ConfBridge(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;2, PA0001,MessageNetConfBridge,MessageNetConfUser) in new stack [Kdevgeis*CLI [0K Channel Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1 was answered. [Kdevgeis*CLI [0K-- Executing [meetme@smvoice-meetme-playfile:1] NoOp(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1, smvoice-meetme-playfile) in new stack -- Executing [meetme@smvoice-meetme-playfile:2] Wait(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1, 1) in new stack [Kdevgeis*CLI [0K-- Executing [meetme@smvoice-meetme-playfile:3] AGI(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1, smvoice,-digium_success,-pa_list) in new stack [Kdevgeis*CLI [0K-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Kdevgeis*CLI [0K == Manager 'MessageNet' logged on from 127.0.0.1 [Kdevgeis*CLI [0K == Manager 'MessageNet' logged off from 127.0.0.1 [Kdevgeis*CLI [0K == Manager 'MessageNet' logged on from 127.0.0.1 [Kdevgeis*CLI [0K == Manager 'MessageNet' logged off from 127.0.0.1 [Kdevgeis*CLI [0K == Using SIP RTP CoS mark 5 [Kdevgeis*CLI [0K --Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1AGI Script smvoice completed, returning 0 [Kdevgeis*CLI [0K-- Executing [meetme@smvoice-meetme-playfile:4] Wait(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1, 1) in new stack [Kdevgeis*CLI [0K Channel SIP/devgeis_to_ebox4300-0004 was answered. [Kdevgeis*CLI [0K-- Executing [smvoice_pa_app_confbridge_intercom@smvoice-transfers:1] ConfBridge(SIP/devgeis_to_ebox4300-0004, PA0001,MessageNetConfBridge,MessageNetConfUser) in new stack [Kdevgeis*CLI [0K-- Executing [meetme@smvoice-meetme-playfile:5] AGI(Local/app_confbridge_call_out@smvoice-local-public-address-playfile-5229;1, smvoice,-digium_success,-pa_single) in new stack [Kdevgeis*CLI [0K-- Launched AGI Script /var/lib/asterisk/agi-bin/smvoice [Kdevgeis*CLI [0K == Manager 'MessageNet' logged on from 127.0.0.1
Re: [asterisk-users] Check for the voicemail
Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. regards, Ruben Am 21.08.2012 19:20, schrieb Danilo Dionisi: I'm sorry, how i can to check for the return code of the smtp session? I've never done :p Thanks, Danilo Il 21/08/12 19:05, Ruben Rögels ha scritto: Okay, so have a look at mailcmd= option in voicemail.conf mailbox will mean a e-mail-box in the next lines. What you need to do is wirting a shell script or what ever to check for the return code of the smtp session (normally it should be a 450 in case of full mailbox). In case of 450 mailbox full whatsoever you need to create a call file in /var/spool/asterisk/outgoing directing your recipient phone to a special extension which handles the playback of the your mailbox is full message. But: this will only work if you deliver the mail directly to the recipients mail server, if you use a smart host, it will accept the message and the final recipient server will bounce the mail back to you. I hope this helps. regards, Ruben Am 21.08.2012 18:52, schrieb Danilo Dionisi: I'm sorry, I haven't been clear. I do not have to check the inbox on Asterisk, but I have to check the free space on a particular mailbox of Exchange software. It's possible with the pair Asterisk-Sendmail? Il 21/08/12 18:45, Danny Nicholas ha scritto: Assuming that you are using the standard 100 message limit, just check for INBOX/MSG0100.txt and send the message. -- _ -- 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] Check for the voicemail
On 21 Aug 2012, at 19:32, Ruben Rögels ruben.roeg...@jumping-frog.org wrote: Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. regards, Ruben It is not very difficult to write an ad-hoc script in a language like Python and call it instead of the regular sendmail command. Just look up something 'Python smtp send tutorial' and you should get a good starting point. Regards, Aldo Sent from my iPhone -- _ -- 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 queue calls - emulate Dial(b) functionality
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Richard Mudgett Sent: Monday, August 20, 2012 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 11 queue calls - emulate Dial(b) functionality I currently run an Asterisk 10 system with hotdesking functionality set up. Several of the users have worked with a system in the past that supported BLF on their IP phones, and would like their current phones to behave in a similar fashion. Right now I have a really kludgy system that mostly works, but doesn’t consistently trigger the cleanup macro to “clear” the device state on the end of a call. Rather than continue to beat my head against the wall playing “which context isn’t firing an h extension to dump calls into the cleanup macro”, I decided to investigate Asterisk 11 for the new Dial() b function and the new hangup handler CHANNEL variable. I have the hints working more or less correctly on direct calls to/from the phones, making use of the b and U functions in Dial() and some judicious use of GROUP channel variables and CHANNEL(hangup_handler_wipe). But, on my live system, sometimes the users receive calls from a queue, and I don’t see any way with the queue calls to emulate the b functionality in Dial() to be able to set the agent extension’s device state to RINGING when the queue call gets created. Obviously, I can use membergosub to set the agent to “INUSE” after they pick up the call (like Dial() U), but is there anything that I can use to manipulate the channel that is calling the agent while/before it is ringing? You could use local channels as queue members. Then you can use Dial(b) when the call goes out to the actual extension. Richard -- Heh, didn't really think of that. It looks like that should do what I need it to. Thanks. Thank you, Noah Engelberth MetaLINK Technologies -- _ -- 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 - BLF on Custom devices
I'd do a packet capture -- ideally from the phone, or using your switch to mirror the phone's port -- and look for a SIP NOTIFY. Then we can know if a NOTIFY is not being sent, or if it's just not being processed as desired by your Cisco SPA 509G. If it's not there, do the same on your asterisk server, and if we see it on the asterisk server but not at the endpoint, we can suspect network configuration. You can also get some more detail about what endpoints are subscribed with sip show subscriptions in the asterisk console, but since it says Watchers 2, that suggests the subscription has been made. I'd just verify that specifically the device you are using for testing is subscribed. Also, I know if you turn the verbosity up high enough (core set verbosity 3) you will get messages in the console about notifications sent. You can also set sip debug on or something along those lines and have Asterisk print all the SIP traffic it's attempting to send. Should help you narrow the possible causes. Here's the sequence I'm seeing with a packet capture from the Asterisk server: - 301 calls 302. The INVITE from 301 hits Asterisk. Asterisk sends back 100 Trying, and then a NOTIFY to 302 and 303 (which are both subscribed to 301, and they see 301 change to in use). 302 and 303 send back 200 OK for the NOTIFYs. - Asterisk sends the INVITE to 302. 302 sends back 100 Trying and then 180 Ringing. After the 180 Ringing, Asterisk sends a NOTIFY to 301 (which I have subscribed to hint 305, which is mapped to SIP/device corresponding to the one I have logged in as 302). No other NOTIFY updates are sent at this time. - I pick up on 302. 302 sends a 200 OK with session description to Asterisk. Asterisk ACKs this 200 and then sends 2 NOTIFYs to 301 (which is subscribed to both the 302 Custom device hint and the 305 SIP Physical device hint), and also sends 1 NOTIFY to 303 (which is only subscribed to the 302 Custom device hint). - While 301 and 302 are still on the call, 303 calls 302. The INVITE from 303 hits Asterisk. Asterisk sends back 100 Trying, and then a NOTIFY to 302 (subscribed to the Custom device hint for 303) and 2 NOTIFYs to 301 (subscribed to both the Custom device and SIP Physical device hints). - Asterisk sends the INVITE to 302. 302 sends back 100 Trying and then 180 Ringing. After the 180 Ringing, Asterisk sends a NOTIFY to 301 (which updates the SIP Physical device hint). No other NOTIFY updates are sent at this time. - 303 cancels the call and sends a CANCEL to Asterisk. Asterisk responds with 200 OK, sends a NOTIFY to 301, sends a CANCEL to 302, and then sends another NOTIFY to 301. 303 then ACKs Asterisk's SIP 487, and Asterisk sends 1 additional NOTIFY to 302 and 301. - 301 and 302 finish their call. Asterisk sends a total of 1 NOTIFY to 302 (the status for Custom device 301), 2 NOTIFYs to 303 (the status for Custom device 301 and 302), and 2 NOTIFYs to 301 (the status for Custom device 302 and Physical Device 302). So, it looks to me like I'm missing NOTIFYs for RINGING or RING_INUSE events on Custom devices. (Slightly sanitized) Verbose 3 output is pastebin'd: http://pastebin.com/q51XUeHe The short of the output is - there is no console output showing == Extension Changed 302[hints] new state on the Ringing or InUseRinging events - only on InUse or Idle events (which matches what I'm seeing on the phones). Thank you, Noah Engelberth MetaLINK Technologies -- _ -- 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 - BLF on Custom devices
On 08/21/2012 02:52 PM, Noah Engelberth wrote: The short of the output is -- there is no console output showing == Extension Changed 302[hints] new state on the Ringing or InUseRinging events -- only on InUse or Idle events (which matches what I'm seeing on the phones). Weird. I just did a test on my production system, 1.8.11.1 (from digium repository) on Debian Squeeze and a Snom 870 running firmware 8.4.35. I had the verbosity set to 3, and I manually changed the devstate in the console with devstate change 207-support-agent RINGING. I got a message: Extension Changed 207-support-agent[employees] new state Ringing for Notify User pfrost and my phone updated its interface to reflect the change. Tested also with INUSE, NOT_INUSE, and RINGINUSE and all resulted in a NOTIFY being sent to the handset. Though the Snom 870 doesn't distinguish between RING and RINGINUSE, it did change from displaying talking to idle or ringing. So, at least with my environment, asterisk is capable and successful at sending SIP NOTIFY for custom devstates. I've no idea why you'd get updates for NOT_INUSE and INUSE but none of the other states. It looks like your dialplan logic is complex enough that I'm having a hard time following it and understanding what should be happening, so I'd suggest testing twiddling the device states through the console and see if that generates NOTIFYs. If not, I'd verify that something is actually subscribed with sip show subscriptions. If something is subscribed and a NOTIFY isn't generated, I'd suspect a bug. If you do get NOTIFYs when changing the devstate in the console but not when making calls, then maybe something in your dialplan logic is wrong, and I'd say simplify it until the problem is found, or it's simple enough to re-ask on the list. -- _ -- 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 - BLF on Custom devices
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost Sent: Tuesday, August 21, 2012 3:20 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 11 - BLF on Custom devices On 08/21/2012 02:52 PM, Noah Engelberth wrote: The short of the output is - there is no console output showing == Extension Changed 302[hints] new state on the Ringing or InUseRinging events - only on InUse or Idle events (which matches what I'm seeing on the phones). Weird. I just did a test on my production system, 1.8.11.1 (from digium repository) on Debian Squeeze and a Snom 870 running firmware 8.4.35. I had the verbosity set to 3, and I manually changed the devstate in the console with devstate change 207-support-agent RINGING. I got a message: Extension Changed 207-support-agent[employees] new state Ringing for Notify User pfrost and my phone updated its interface to reflect the change. Tested also with INUSE, NOT_INUSE, and RINGINUSE and all resulted in a NOTIFY being sent to the handset. Though the Snom 870 doesn't distinguish between RING and RINGINUSE, it did change from displaying talking to idle or ringing. So, at least with my environment, asterisk is capable and successful at sending SIP NOTIFY for custom devstates. I've no idea why you'd get updates for NOT_INUSE and INUSE but none of the other states. It looks like your dialplan logic is complex enough that I'm having a hard time following it and understanding what should be happening, so I'd suggest testing twiddling the device states through the console and see if that generates NOTIFYs. If not, I'd verify that something is actually subscribed with sip show subscriptions. If something is subscribed and a NOTIFY isn't generated, I'd suspect a bug. If you do get NOTIFYs when changing the devstate in the console but not when making calls, then maybe something in your dialplan logic is wrong, and I'd say simplify it until the problem is found, or it's simple enough to re-ask on the list. Changing the Custom devstate from the CLI generates Extension Changed for Notify User messages when I change to INUSE or NOT_INUSE, but not when I change to RINGING or RINGINUSE. If I change from NOT_INUSE to RINGING and then back to NOT_INUSE, no notify messages are being generated. core show hints does show the correct state for 301@hints when I make the manual change to RINGINUSE (or RINGING). sip show subscriptions does show 2 subscriptions (1 each from 302 and 303) for 301@hints. Thank you, Noah Engelberth MetaLINK Technologies -- _ -- 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] Asterisk 11 - XMPP and JabberSend()
I'm trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). My jabber.conf file is as follows: [general] debug=no autoprune=no [testaccount] type=client serverhost=my.jabber.server username=myaccount@my.jabber.server secret=mypassword port=jabberport usetls=yes usesasl=yes xmpp show connections gives the following output from the console: testasterisk11*CLI xmpp show connections Jabber Users and their status: [testaccount] aster...@jabber.metalink.net - Connected Number of clients: 1 xmpp show buddies lists out the users that are being auto-added to the buddy list from the XMPP server. I try to have a test extension send a message, and get this output (and the call fails with a Declined message on the calling phone). -- Executing [20005@metalink:2] JabberSend(SIP/649EF376CA25-000c, testaccount,user@my.jabber.server,Test) in new stack [Aug 21 15:42:54] WARNING[20469][C-000c]: res_xmpp.c:1752 xmpp_send_exec: JabberSend requires arguments (account,jid,message) I've tried putting in the full username and the [testaccount] for the first argument to JabberSend. I've tried just the username as well as the full user@my.jabber.servermailto:user@my.jabber.server in the jid argument. I've tried multi-word and single word messages. I've tried encapsulating each argument in quotes. With XMPP debug on, I can see presence messages being received XMPP received from 'test account', but no XMPP debug output occurs when I try to place my test call to send the XMPP message. Kinda going cross-eyed from looking at this - is there anything else I should try or anything wrong in my configuration? Thank you, Noah Engelberth MetaLINK Technologies -- _ -- 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 - XMPP and JabberSend()
- Original Message - I’m trying to get my Asterisk 11 test box set up with XMPP, having troubles with JabberSend(). Hola! The underlying issue here (finger didn't hit shift to turn 2 into @) has been fixed in Asterisk 11 as of revision 371518. You can grab that specific fix from subversion, grab Asterisk 11 from subversion, or wait for the next beta. It's a minor change. Sorry for the inconvenience! Cheers, -- Joshua Colp Digium, Inc. | Senior Bacon Inspector 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Issue.
Up? 2012/8/20 Luis H. Forchesatto luisforchesa...@gmail.com Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton rnew...@digium.com On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via the Overtek ATA the DTMF works fine but at the others ATA it doesn't. My config: - asterisk 1.6.2.13 - dahdi 2.3.0.1 - The phones connected are all physical phones There is additional data you can provide to make it easier for others to help out: If you can pastebin an Asterisk log including all message types plus VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would be very helpful. A step beyond that is to also provide a SIP and RTP packet trace so that whoever wants to help can look through it in Wireshark. If you can get the packet trace for the same calls you gather log data for, that would be best. Thanks! [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+** Debug+Informationhttps://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information -- Rusty Newton Digium, Inc | Open Source Community Support Manager Check us out at: www.digium.com www.asterisk.org -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Att.* *** -- Att.* *** -- _ -- 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 - BLF on Custom devices
She's talking about asterisk 11 not asterisk 1.8.11 -Original Message- From: Phil Frost p...@macprofessionals.com Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 21 Aug 2012 15:19:31 To: asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 11 - BLF on Custom devices -- _ -- 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] comma issue with func_odbc
Hey all
I have an issue that I have been bumping up against. We have some inbound
fax services and occasionally an inbound fax that successfully came in
would fail to store it's references in the database.
We are using a function in func_odbc to update a database table. We call
the function from the dialplan and pass in all the opt_xxx return values as
well as other important values we need to store. The issue we are having is
this.
In some cases values such as Caller ID, Remote Station ID and Header Info
will have a comma in them. Even though these values are being encoded using
SQL_ESC the comma is being interpreted as an extra parameter and messing up
the storage values and causing the database insert to fail. Is there a way
to encode variables with commas in their values so they can be sent into
the func_odbc function as a parameter without causing the process to thing
it has more parameters?
Exp...
If the ${CALLERID(number)} or ${FAXOPT(remotestationid)} or others in the
line below have a comma in them then the parameter order pushed buy one and
the value is broken up when building the insert statement.
exten =
Do-Store,n,Set(FAX-DO-STORE()=${CALLERID(number)},${CALLERID(name)},${l_faxF
ile_Path},${l_faxFile_FullName},${FAXOPT(ecm)},${FAXOPT(filename)},${FAXOPT(
localstationid)},${FAXOPT(headerinfo)},${FAXOPT(remotestationid)},${FAXOPT(m
axrate)},${FAXOPT(minrate)},${l_storeRate},${FAXOPT(pages)},${FAXOPT(resolut
ion)},${FAXOPT(error)},${FAXOPT(status)},${FAXOPT(statusstr)})
Any ideas suggestions on how to over come this. Would be appreciated.
Thanks
Bryant Any ideas suggestions on how to over come this. Would be
appreciated.ThanksBryant
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Re: [asterisk-users] comma issue with func_odbc
I have an issue that I have been bumping up against. We have some
inbound fax services and occasionally an inbound fax that
successfully came in would fail to store it's references in the
database.
We are using a function in func_odbc to update a database table. We
call the function from the dialplan and pass in all the opt_xxx
return values as well as other important values we need to store.
The issue we are having is this.
In some cases values such as Caller ID, Remote Station ID and Header
Info will have a comma in them. Even though these values are being
encoded using SQL_ESC the comma is being interpreted as an extra
parameter and messing up the storage values and causing the database
insert to fail. Is there a way to encode variables with commas in
their values so they can be sent into the func_odbc function as a
parameter without causing the process to thing it has more
parameters?
Exp...
If the ${CALLERID(number)} or ${FAXOPT(remotestationid)} or others in
the line below have a comma in them then the parameter order pushed
buy one and the value is broken up when building the insert
statement.
exten =
Do-Store,n,Set(FAX-DO-STORE()=${CALLERID(number)},${CALLERID(name)},${l_faxFile_Path},${l_faxFile_FullName},${FAXOPT(ecm)},${FAXOPT(filename)},${FAXOPT(localstationid)},${FAXOPT(headerinfo)},${FAXOPT(remotestationid)},${FAXOPT(maxrate)},${FAXOPT(minrate)},${l_storeRate},${FAXOPT(pages)},${FAXOPT(resolution)},${FAXOPT(error)},${FAXOPT(status)},${FAXOPT(statusstr)})
Any ideas suggestions on how to over come this. Would be appreciated.
Can you quote it like this:
DO-STORE()=${CALLERID(number)},${CALLERID(name)},...
Richard
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[asterisk-users] recording calls
I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /719000 -- SIP/BVTrunk-0163 is making progress passing it to SIP/1010-0162 -- SIP/BVTrunk-0163 answered SIP/1010-0162 -- Feature Found: apprecord exten: apprecord -- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0162, 0?Set(THISEXTEN=719)) in new stack -- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0162, 1?Set(THISEXTEN=1010)) in new stack -- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0162, 0?MacroExit()) in new stack -- Executing [s@macro-one-touch-record:4] GotoIf(SIP/1010-0162, 0?stoprec) in new stack -- Executing [s@macro-one-touch-record:5] GotoIf(SIP/1010-0162, 0?stopped) in new stack -- Executing [s@macro-one-touch-record:6] GotoIf(SIP/1010-0162, 0?recording) in new stack -- Executing [s@macro-one-touch-record:7] Set(SIP/1010-0162, MASTER_CHANNEL(ONETOUCH_REC)=RECORDING) in new stack -- Executing [s@macro-one-touch-record:8] Set(SIP/1010-0162, MASTER_CHANNEL(REC_STATUS)=RECORDING) in new stack -- Executing [s@macro-one-touch-record:9] Set(SIP/1010-0162, AUDIOHOOK_INHERIT(MixMonitor)=yes) in new stack -- Executing [s@macro-one-touch-record:10] MixMonitor(SIP/1010-0162, 2012/08/21/out-719000-1010-20120821-183119-1345595479.530.wav,a,) in new stack == Begin MixMonitor Recording SIP/1010-0162 -- Executing [s@macro-one-touch-record:11] Set(SIP/1010-0162, MON_FMT=wav) in new stack -- Executing [s@macro-one-touch-record:12] Set(SIP/1010-0162, MASTER_CHANNEL(CDR(recordingfile))=out-719000-1010-20120821-183119-1345595479.530.wav) in new stack -- Executing [s@macro-one-touch-record:13] Set(SIP/1010-0162, MASTER_CHANNEL(ONETOUCH_RECFILE)=out-719000-1010-20120821-183119-1345595479.530.wav) in new stack -- Executing [s@macro-one-touch-record:14] Playback(SIP/1010-0162, beep) in new stack -- SIP/1010-0162 Playing 'beep.ulaw' (language 'en') -- Executing [s@macro-one-touch-record:15] MacroExit(SIP/1010-0162, ) in new stack -- Executing [h@macro-dialout-trunk:1] Macro(SIP/1010-0162, hangupcall,) in new stack -- Executing [s@macro-hangupcall:1] GotoIf(SIP/1010-0162, 1?theend) in new stack -- Goto (macro-hangupcall,s,3) -- Executing [s@macro-hangupcall:3] ExecIf(SIP/1010-0162, 1?Set(CDR(recordingfile)=out-719000-1010-20120821-183119-1345595479.530.wav)) in new stack -- Executing [s@macro-hangupcall:4] Hangup(SIP/1010-0162, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/1010-0162' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/1010-0162' == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/1010-0162' in macro 'dialout-trunk' == Spawn extension (from-internal, 719000, 6) exited non-zero on 'SIP/1010-0162' == MixMonitor close filestream == End MixMonitor Recording SIP/1010-0162 == Extension Changed 1010[ext-local] new state Idle for Notify User 1004 On inbound calls I see: == Using SIP RTP CoS mark 5 -- Called SIP/1010 -- Connected line update to SIP/ BVTrunk -0160 prevented. == Extension Changed 1010[ext-local] new state Ringing for Notify User 1004 -- SIP/1010-0161 is ringing -- Connected line update to SIP/ BVTrunk -0160 prevented. -- SIP/1010-0161 answered SIP/ BVTrunk -0160 == Extension Changed 1010[ext-local] new state InUse for Notify User 1004 -- Executing [s@macro-auto-blkvm:1] Set(SIP/1010-0161, __MACRO_RESULT=) in new stack -- Executing [s@macro-auto-blkvm:2] Macro(SIP/1010-0161, blkvm-clr,) in new stack -- Executing [s@macro-blkvm-clr:1] Set(SIP/1010-0161, SHARED(BLKVM,SIP/BVTrunk-0160)=) in new stack -- Executing [s@macro-blkvm-clr:2] Set(SIP/1010-0161, GOSUB_RETVAL=) in new stack -- Executing [s@macro-blkvm-clr:3] MacroExit(SIP/1010-0161, ) in new stack -- Executing [s@macro-auto-blkvm:3] ExecIf(SIP/1010-0161, 0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=1010)) in new stack -- Executing [s@macro-auto-blkvm:4] ExecIf(SIP/1010-0161, 0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Josh Hopkins)) in new stack -- Feature Found: apprecord exten: apprecord -- Executing [s@macro-one-touch-record:1] ExecIf(SIP/1010-0161, 0?Set(THISEXTEN=1010)) in new stack -- Executing [s@macro-one-touch-record:2] ExecIf(SIP/1010-0161, 1?Set(THISEXTEN=1010)) in new stack -- Executing [s@macro-one-touch-record:3] ExecIf(SIP/1010-0161, 1?MacroExit()) in new stack
Re: [asterisk-users] Check for the voicemail
On Tuesday 21 Aug 2012, Ruben Rögels wrote: Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. Something like this ought to do it: (sleep 5; echo HELO foo; sleep 1; \ echo mail from: f...@example.com; sleep 1; \ echo rcpt to: userid.t...@youwant.to.check; sleep 1; \ echo data; echo test; echo .; sleep 1; echo quit) | \ telnet mail.ho.st 25 21 | fgrep -q '450 5.7.1' notify-user.sh Of course, it's probably better to wrap this into a Perl or equivalent script, but it should work on the shell too. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Check for the voicemail
I would simply send the message with sendmail -v and then grep the output for the error message Il giorno 22/ago/2012 04:19, Raj Mathur (राज माथुर) r...@linux-delhi.org ha scritto: On Tuesday 21 Aug 2012, Ruben Rögels wrote: Hello, no problem at all, I think this is the tricky part. A smtp dialogue between your email client and a smtp server normally looks like this: user@box:~? netcat mx1.example.com 220 postfix ESMTP mx1.example.com helo me.local 250 mx1.example.com mail from: ruben.roeg...@wiseape.de 250 2.1.0 Ok rcpt to: ruben.roeg...@example.com 450 5.7.1 ruben.roeg...@example.com: Mailbox Full The tricky part is writing or finding a console smtp client that gives you feedback about the 450 error that just happened. Right now I cannot give you a precise way to do that, but I have basic understanding of the technology, so I know that it is possible to do so ;-) I'm looking around in the net, because I think I'll soon have to handle your problem aswell in my company ;-) If I can find solution, I'll post it. Something like this ought to do it: (sleep 5; echo HELO foo; sleep 1; \ echo mail from: f...@example.com; sleep 1; \ echo rcpt to: userid.t...@youwant.to.check; sleep 1; \ echo data; echo test; echo .; sleep 1; echo quit) | \ telnet mail.ho.st 25 21 | fgrep -q '450 5.7.1' notify-user.sh Of course, it's probably better to wrap this into a Perl or equivalent script, but it should work on the shell too. Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Load test for FXS and FXO cards
Hi, Can anyone tell me how to do the load test for the FXS and FXO cards and find how much the asterisk machine can load for different processors configuration . Regards Upendra. -- _ -- 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] Check for the voicemail
On Wednesday 22 Aug 2012, Roberto Piola wrote: I would simply send the message with sendmail -v and then grep the output for the error message Er, that works too :) Much better solution (as long as you are root). Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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
