Re: [asterisk-users] Unable to load cdr_adaptive_odbc.so
On Wednesday 21 April 2010 17:11:38 Alejandro Recarey wrote: Thanks Tilghman, this immediatley solved the problem. Perhaps a mention in cdr_adaptive_odbc.conf that the res_odbc.so module must also be loaded will help newbies like me ;) In general, it's a good idea to load all modules that are built, unless you have a specific need to ensure that certain modules are not loaded (such as on embedded systems). If you do want to disable modules, disabling them in 'make menuselect' will ensure that any dependencies of those modules will also be immediately disabled. For example, when highlighting cdr_adaptive_odbc in menuselect, it indicates that res_odbc is a dependency; disabling res_odbc will also disable cdr_adaptive_odbc, func_odbc, and several other modules that require res_odbc to function properly. Consequently, re-enabling cdr_adaptive_odbc will also re-enable res_odbc. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) 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] Time difference in CSV CDR's and MySQL CDR's
the time in the file cdr is right, as mysql. calldate is the time when the record insert into mysql. 2010/4/22 Alejandro Recarey alexreca...@gmail.com: Hi all, I am having a curious problem. I use two cdr backends, csv and MySQL. These are my settings: Call Detail Record (CDR) settings -- Logging: Enabled Mode: Batch Log unanswered calls: Yes * Batch Mode Settings --- Safe shutdown: Enabled Threading model: Scheduler plus separate threads Current batch size: 0 records Maximum batch size: 25 records Maximum batch time: 10 seconds Next batch processing time: 7 seconds * Registered Backends --- csv mysql cdr-custom I am finding that the calldate field varies between 3 seconds and 3 minutes between the MySQL database and the CSV files! Is this expected behaviour? I thought they should both use the same timestamp. Is is very difficult to match CDR's this way, and I am finding it hard to trust the results, as I wanted to make sure that my database was behaving correctly and not losing any CDR's along the way. Which one of the two CDR's is correct? Should this be posted as a bug? Regards, Alex -- _ -- 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 -- Best regards, Sucan -- _ -- 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] Need to patch Asterisk for problem with FreePBX Call Confirmation
I have installed a fresh installation of AsteriskNOW and have configured FreePBX. When my users receive a call to their extension the Follow Me rules call their cell phone. I currently have Call Confirmation enabled. When the user attempts to press 1 to accept the call they are immediately disconnected. I have found that this is a bug with Asterisk. The bug is reported here: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=14940 I see that a patch has been released to fix this issue but I can not figure out how to patch my AsteriskNOW distribution. I am wondering if someone could provide some instructions on how to patch this issue. This is causing major problems for my users. Asterisk Version 1.4.24 FreePBX 2.6.0.2 Thanks ___ Just ask for ASK Taking the hassle out of technology so you can run your business. 1-877-ASK-4-ASK Did you know you can chat online with us? http://www.justask.net/support Follow ASK on twitter at: http://twitter.com/justasknet-- _ -- 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] Portech MV-374 does not register behind NAT
When I comment out the port-parameter (then it defaults to 5060), it is still the same... [Apr 22 09:32:49] --- Transmitting (NAT) to my_pub_ip:5064 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.23:5064;branch=z9hG4bKc46696a2b5;received=my_pub_ip From: "SIM 3-1" sip:simsim@pub_ast_server;tag=0a99a41b To: "SIM 3-1" sip:sim...@pub_ast_server;tag=as461d5769 Call-ID: 5c580e091901a03c39becff71e477...@192.168.1.23 CSeq: 89 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="103001vc", nonce="03e68412" Content-Length: 0 Jonas. bruce bruce wrote: Try changing port=5064 to port=5060 in your Asterisk config file. Portech will negotiate it's port with Asterisk itself. -- _ -- 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] DAHDI User-User information Message longer than it should be??
Hi. My configuration is Elastix 1.5.2-2 (asterisk 1.4.24, libpri-1.4.3-5, dahdi-2.1.0.4-7 ) and OpenVox d210e connected to telco provider (Euro ISDN). Here is my /etc/dahdi/system.conf: # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS ClockSource span=1,1,0,ccs,hdb3 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=oslec,1-15,17-31 # Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED span=2,2,0,ccs,hdb3 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=oslec,32-46,48-62 # Global data loadzone= ru defaultzone = ru ... and /etc/asterisk/chan_dahdi.conf: [trunkgroups] [channels] context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 ;Uncomment these lines if you have problems with the disconection of your analog lines ;busydetect=yes ;busycount=3 immediate=no ; Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS ClockSource group=0,11 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 ; Span 2: TE2/0/2 T2XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED group=0,12 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 32-46,48-62 context = default group = 63 On incoming call from telco I'm getting this (pri debug, full log attached): Protocol Discriminator: Q.931 (8) len=39 Call Ref: len= 2 (reference 538/0x21A) (Originator) Message type: SETUP (5) [a1] Sending Complete (len= 1) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 9a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 26 ] [6c 09 41 81 32 37 39 39 30 39 39] Calling Number (len=11) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '3800100' ] [70 06 c1 36 39 34 31 31] Called Number (len= 8) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '70522' ] [7e 04 00 09 f3 63] User-User Information (len= 6) [ 00 09 73 63 ] [Apr 22 11:25:15] ERROR[29838]: chan_dahdi.c:9482 dahdi_pri_error: XXX Message longer than it should be?? XXX -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) [Apr 22 11:25:15] ERROR[29838]: chan_dahdi.c:9482 dahdi_pri_error: XXX Message longer than it should be?? XXX Sending Receiver Ready (86) After many attempts it gives up and sending me DISCONNECT: Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 539/0x21B) (Originator) Message type: DISCONNECT (69) [08 02 80 e6] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Recover on timer expiry (102), class = Protocol Error (e.g. unknown message) (6) ] On this message DAHDI replies with RELEASE COMPLETE: Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 539/0x21B) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 81 d1] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Invalid call reference value (81), class = Invalid message (e.g. parameter out of range) (5) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate == I have another asterisk box which works with this telco. My configuration there is rather different in both hardware and software (no PCI-E and no Elastix there, plain asterisk 1.4.21.2 built from source, libpri 1.4.4 and zaptel 1.4.11). That fact causes me to think that smth wrong with my DAHDI configuration. Any ideas? -- Alexandr Krylovskiy -- _ -- 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
Re: [asterisk-users] Why app_fax.so there is no in asterisk16-1.6.2.6-1_centos5.x86_64.rpm?
http://packages.asterisk.org/centos/5/current/x86_64/RPMS/
On 21.04.2010 17:34, David Backeberg wrote:
I didn't know there was an RPM for centos with asterisk in it.
I personally think that's a bad idea. There are a lot of source options.
app_fax.so in particular depends on SpanDSP, and particular versions thereof.
That's probably why it's missing from somebody's RPM.
Build from source.
On Wed, Apr 21, 2010 at 7:02 AM, Самусенко Андрейsamuse...@msm.ru wrote:
1. Subject.
2. asterisk16-1.6.2.6-1_centos5.src.rpm have not asterisk.logrotate in
SOURCES
3. for --without dahdi
diff SPECS/asterisk16-my.spec SPECS/asterisk16.spec
750a750
%{_libdir}/asterisk/modules/res_timing_dahdi.so
879d878
%{_libdir}/asterisk/modules/res_timing_dahdi.so
--
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Re: [asterisk-users] Time difference in CSV CDR's and MySQL CDR's
Un-top-posting... 2010/4/22 Alejandro Recarey alexreca...@gmail.com: I am having a curious problem. I use two cdr backends, csv and MySQL. I am finding that the calldate field varies between 3 seconds and 3 minutes between the MySQL database and the CSV files! Is this expected behaviour? I thought they should both use the same timestamp. On Thu, 22 Apr 2010, Zhang Shukun wrote: the time in the file cdr is right, as mysql. calldate is the time when the record insert into mysql. I'm just a 1.2 Luddite, but... In cdr_addon_mysql.c: localtime_r(cdr-start.tv_sec,tm); strftime(timestr,128,DATE_FORMAT,tm); and then timestr is used to populate the 'calldate' column when the insert statement is built. Which is consistent with my CDRs -- they show the time the call was started, not some time after the call is finished when the row is inserted into the database. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Improving audio bitrate for all callers in a conference room for a podcast
Hi! Is there any way to configure a stock Asterisk install to use wideband mixing or will we have to compile our own? Not sure! Look here: http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe#ConfBridge Philipp -- _ -- 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] Odd Issue With Polycom Phones
I appears as though I was a little hasty in saying that it wasn't generating two calls. It actually was, but I was doing a poor job of searching the logs. I setup a new-to-me IP 6000 with older firmware on it (3.0.2.0927), and I am not getting the issue. I am going to start upgrading the firmware on that phone to see when it breaks, then it looks like I will call Polycom to try to get them to fix the issue. Thanks for the help from everyone on this. -Jay From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sean Brady Sent: Wednesday, April 21, 2010 5:49 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Odd Issue With Polycom Phones On 04/21/2010 03:08 PM, Warren Selby wrote: On Wed, Apr 21, 2010 at 3:46 PM, Jay Vocaire jvoca...@innproc.commailto:jvoca...@innproc.com wrote: Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the tag= determines the call. The first time it sends like this: snip Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: snip The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username=3271, realm=asterisk, nonce=393a1b1f, uri=sip:3...@y.y.y.y;user=phonesip:3...@y.y.y.y;user=phone, response=c8223e261c252c12172982ee661ad307, algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. This is expected behavior for SIP communications. I see this all the time when an end point is registering with Asterisk. I think in those cases, however, it's a REGISTER request, not an INVITE. How is your sip.conf configured for these end points? Do you have any phones other than the ones experiencing this problem that you can test with? Yes this is expected behavior on a REGISTER. I didn't think that it was correct on an INVITE, however on reading RFC 3261, I believe that Asterisk is correctly responding to the request, needing credentials from the UA (Polycom). My Ekiga softphone is doing the exact same thing, however it's not creating the same 2 call issue that your Polycoms are having. The Ekiga call setup is not including credentials on the first INVITE, receives a 401 not authorized, and sends another INVITE with credentials, and receives a 100 TRYING from Asterisk. This is most likely an issue with the firmware on the Polycom. Bottom line is that another UA is doing the same thing, the call is setup properly, and it appears to work. I respectfully request that someone smarter than me take a look at this and verify my conclusions, or correct me accordingly. Thanks. According to RFC 3261 (note that the RFC uses the word request instead of register or registration request): ... If a 401 (Unauthorized) or 407 (Proxy Authentication Required) response is received, the UAC SHOULD follow the authorization procedures of Section 22.2 and Section 22.3 to retry the request with credentials. ... Read more: http://www.faqs.org/rfcs/rfc3261.html#ixzz0llyASXyI ... 22.2 User-to-User Authentication When a UAS receives a request from a UAC, the UAS MAY authenticate the originator before the request is processed. If no credentials (in the Authorization header field) are provided in the request, the UAS can challenge the originator to provide credentials by rejecting the request with a 401 (Unauthorized) status code. The WWW-Authenticate response-header field MUST be included in 401 (Unauthorized) response messages. The field value consists of at least one challenge that indicates the authentication scheme(s) and parameters applicable to the realm. An example of the WWW-Authenticate header field in a 401 challenge is: WWW-Authenticate: Digest realm=biloxi.com, qop=auth,auth-int, nonce=dcd98b7102dd2f0e8b11d0f600bfb0c093, opaque=5ccc069c403ebaf9f0171e9517f40e41 When the originating UAC receives the 401 (Unauthorized), it SHOULD, if it is able, re-originate the request with the proper credentials. The UAC may require input from the originating user before proceeding. Once authentication credentials have been supplied (either directly by the user, or discovered in an internal keyring), UAs SHOULD cache the credentials for a given value of the To header field and realm and attempt to re-use these values on the next request for that destination. UAs MAY cache credentials in any way they would like. If no credentials for a realm can be located, UACs MAY attempt to retry the request with a username of anonymous and no password (a password of ). Once credentials have been located, any UA that wishes to authenticate itself with a UAS or registrar -- usually,
Re: [asterisk-users] Portech MV-374 does not register behind NAT
Try reseting the Gateway (soft reset of the settings) and use only IE to do the setup again. Nothing else comes to my mind. Also, create a simple extension in Asterisk or if you are using FreePBX you don't need to tamper with any ports stuff. -Bruce On Thu, Apr 22, 2010 at 3:37 AM, Jonas Kellens jonas.kell...@telenet.bewrote: When I comment out the port-parameter (then it defaults to 5060), it is still the same... [Apr 22 09:32:49] --- Transmitting (NAT) to my_pub_ip:5064 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.23:5064;branch=z9hG4bKc46696a2b5;received= my_pub_ip From: SIM 3-1 sip:sim...@pub_ast_server;tag=0a99a41b To: SIM 3-1 sip:sim...@pub_ast_server;tag=as461d5769 Call-ID: 5c580e091901a03c39becff71e477...@192.168.1.23 CSeq: 89 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm=103001vc, nonce=03e68412 Content-Length: 0 Jonas. bruce bruce wrote: Try changing port=5064 to port=5060 in your Asterisk config file. Portech will negotiate it's port with Asterisk itself. -- _ -- 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] Portech MV-374 does not register behind NAT
I'm using Firefox on Fedora but I don't think the problems lies there. All goes well when the gateway is connected directly to the internet... It's when it is behind NAT the 401 is sent from Asterisk... It must be some NAT-thing combination in how the GSM-gateway/Zyxel-router sends the registration... Nobody on this list has experienced similar issues ?? Jonas. bruce bruce wrote: Try reseting the Gateway (soft reset of the settings) and use only IE to do the setup again. Nothing else comes to my mind. Also, create a simple extension in Asterisk or if you are using FreePBX you don't need to tamper with any ports stuff. -Bruce -- _ -- 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] Need to patch Asterisk for problem with FreePBX Call Confirmation
I was able to upgrade asterisk to 1.4.25 and the issue with Find Me Follow me in FreePBX has been resolved. Thanks *** ___ Just ask for ASK Taking the hassle out of technology so you can run your business. 1-877-ASK-4-ASK Did you know you can chat online with us? http://www.justask.net/support Follow ASK on twitter at: http://twitter.com/justasknet-- _ -- 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] Portech MV-374 does not register behind NAT
On Thu, 2010-04-22 at 17:45 +0200, Jonas Kellens wrote: All goes well when the gateway is connected directly to the internet... It's when it is behind NAT the 401 is sent from Asterisk... Is the device registering to an IP address, or do a DNS name? What type of NAT firewall are you using? This reminds me of a problem I had years ago with a Cisco PIX firewall, where it would rewrite IP addresses in the SIP Request URI, causing the authentication to fail. One solution was to have it register to a fully-qualified domain name instead of an IP address, so that the Request URI wouldn't get overwritten. It's certainly worth a shot... -- Jared Smith Digium, Inc. -- _ -- 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] How to do analog em on asterisk?
Hi, Can anybody with previous experience with it guide me on how to setup asterisk with analog em to connect it to an old style em system which uses 4 pair cables on RJ 45 jacks. All the analog cards I know of use RJ 11 jacks. And there is no choice of modernization of the customer equipment. Cable pin out are as follows: 1. M lead 2. E lead 3. Tip1 4. Ring 5. Tip 6. Ring1 7. SG 8. SB Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. -- _ -- 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] Portech MV-374 does not register behind NAT
Jared, thank you for your answer. As I said in my previous mail, I'm using a Zyxel NBG-419 router (which normally supports VoIP and QoS). Firewall is disabled on the Zyxel. The MV-374 only accepts IP-address, not a FQDN. Will give it another try though... The answer from Portech-support : "use STUN". Even if the NAT rewrites the IP-address/port combination, why is it a problem for the Portech and not for the IP-phones (Grandstream Snom) ? They all communicate on port 5060 -- 5064 (several SIP-accounts) Jonas. Jared Smith wrote: On Thu, 2010-04-22 at 17:45 +0200, Jonas Kellens wrote: All goes well when the gateway is connected directly to the internet... It's when it is behind NAT the 401 is sent from Asterisk... Is the device registering to an IP address, or do a DNS name? What type of NAT firewall are you using? This reminds me of a problem I had years ago with a Cisco PIX firewall, where it would rewrite IP addresses in the SIP Request URI, causing the authentication to fail. One solution was to have it register to a fully-qualified domain name instead of an IP address, so that the Request URI wouldn't get overwritten. It's certainly worth a shot... -- Jared Smith Digium, Inc. -- _ -- 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 choking on voice messages announcements
On 04/21/2010 07:13 PM, bruce bruce wrote: How can I find out what the source of the problem is guys? As I said I didn't change anything, except for making few minor changes to the firewall today and that was at Amazon firewall level and not within CentOS. What causes these bad dahdi_test values? What version of DAHDI are you using? If you're using a relatively recent version (2.2.0+) dahdi_dummy works better on virtual machines because it uses the wall time to adjust itself as opposed to just assuming it's going to be called at a fixed interval. But, one issue with this is that if the host machine is unable to keep accurate wall time then dahdi_dummy can be confused about how much time is really passing. The big jumps in accuracy make me think that NTP may be adjusting the host clock in several second increments? The firewall changes aren't interfering with NTP or anything like that are they? Just a thought... -- 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
[asterisk-users] Follow-me to my answering machine :-(
Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that would be bad (TM). app_followme sounds like a solution. BUT, I also have a car. And I cannot use DTMF to respond to the app_followme prompts (which I WANT, to avoid routing the forwarded call to voicemail when the cell phone is off and its voicemail picks up), while driving. I've tried using dial macros and AMD(), but this is complex, very unreliable, and delays the connection of the call significantly. Is there some way to make app_followme use voice recognition? Or some other solution so that I can get all my phones to ring with one number, even when my cell phone is off or out of range? Bryan Jacobs signature.asc Description: PGP signature -- _ -- 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] Security tests
Hi! But it draws attention to me between the PC with softphone and the telephone I see traffic ARP or ICMP that could make to try between the equipment but does not see RTP. Is there some special consideration that it must to observe? Your English is seriously twisted, making your question impossible to understand. My feeling is that you have used a machine translation service. Your question is probably: I can see ARP and ICMP, but not RTP, what am I missing? How did you place your virtual listening machine into the network, is it connected to an old hub, or a switch, or the mirroring port of a switch, or does it use the same NIC (and computer) as the softphone? You will first need to get in between the two endpoints in order to be able to capture that point-to-point RTP traffic - there are normal and malicious ways to achieve that. Philipp -- _ -- 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] Hangup after n seconds using originate ?
Hi All, I would like to know if you can confirm that, if using origination via AMI, as documented here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate it is not possible to set the max duration of a call. I mean: what you would do with the L (limit) parameter of the command Dial, is not possible when originating. As well as using the absolute timeout, as documented here: http://www.asteriskguru.com/tutorials/timeoutabsolute_function.html can't be done when originating. Is this true ? I'm using version 1.4. Thanks for supporting, have a nice day. Mike -- _ -- 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] Follow-me to my answering machine :-(
Hi, currently I am writing a sound recognition software that will suit here pretty well - it can recognize your cell phone's our of radio coverage or similar operator message. It's GPL, link here: http://github.com/Motiejus/SoundPatty Now the program can say if 2 WAV files match (tested with out of radio coverage status and GSM network - it works), and right now I am working with it's support with asterisk (through JACK_HOOK). It shouldn't take more than a week, I hope. I will announce to this conference when it's ready :) Regards Motiejus On Thu, Apr 22, 2010 at 8:31 PM, Bryan Jacobs n...@landwarsin.asia wrote: Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that would be bad (TM). app_followme sounds like a solution. BUT, I also have a car. And I cannot use DTMF to respond to the app_followme prompts (which I WANT, to avoid routing the forwarded call to voicemail when the cell phone is off and its voicemail picks up), while driving. I've tried using dial macros and AMD(), but this is complex, very unreliable, and delays the connection of the call significantly. Is there some way to make app_followme use voice recognition? Or some other solution so that I can get all my phones to ring with one number, even when my cell phone is off or out of range? Bryan Jacobs -- _ -- 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] Security tests
On Thu, 22 Apr 2010, Philipp von Klitzing wrote: Hi! But it draws attention to me between the PC with softphone and the telephone I see traffic ARP or ICMP that could make to try between the equipment but does not see RTP. Is there some special consideration that it must to observe? Your English is seriously twisted, making your question impossible to understand. My feeling is that you have used a machine translation service. Your question is probably: I can see ARP and ICMP, but not RTP, what am I missing? How did you place your virtual listening machine into the network, is it connected to an old hub, or a switch, or the mirroring port of a switch, or does it use the same NIC (and computer) as the softphone? You will first need to get in between the two endpoints in order to be able to capture that point-to-point RTP traffic - there are normal and malicious ways to achieve that. Depends on what you consider malicious :) ARP Cache poisoning is considered fairly normal by some these days... However the easiest way to capture data is on the asterisk server itself... Gordon -- _ -- 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] Avaya UUI
Hello List, I need to connect with an Avaya PBX (this part is done), and i would like to get and send back User-to-User Information (UUI) with the call. The UUI need because I need to identify the call based on something witch is available on Asterisk and Avaya too. It is possible, or have a better solution? Anybody did it before? Thanks for the help! Zsotya -- _ -- 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] Swaping out phones.
I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload. Any other ideas? Thanks -- _ -- 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] Swaping out phones.
Are you using Databases and/or realtime as opposed to plain-text conf files? If not, I'd try a restart when convenient. If so, something is hung up. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony LaMear Sent: Thursday, April 22, 2010 2:19 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Swaping out phones. I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload. Any other ideas? Thanks -- _ -- 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] Follow-me to my answering machine :-(
Check out the 'p' option for the Dial command. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial It enables call screening, so you have to press 1 to answer. This can also prevent the voice mail from being left on your cell phone. -- _ -- 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] Hangup after n seconds using originate ?
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like that when creating the originate command? I don't know if it works, but it is worth a shot. -- _ -- 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] Hangup after n seconds using originate ?
On Thu, 22 Apr 2010 15:58:34 -0400 Ryan Bullock rrb3...@gmail.com wrote: Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like that when creating the originate command? I don't know if it works, but it is worth a shot. Hi Ryan, thanks for your comment. Unfortunately the 'Variable' parameter is used to push data between the originating script and the dialplan, not commands. Example: Variable: var1=23|var2=24|var3=25 Additionally, this data can be used in the dialplan only when the call gets answered or when it fails. I can't find a way to inject the parameter DURING (or before) the call. Thank you very much for supporting, Mike -- _ -- 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] Hangup after n seconds using originate ?
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel: Local/allow_caller_id:415111:541222:3...@context
Exten: do_echo
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=AllowCallerID
ActionID: AllowCallerID
Async: true
exten = _allow_caller_id.,1,Verbose(1,allow_caller_id gets ${EXTEN})
exten = _allow_caller_id.,n,Set(MyCallerID=${CUT(EXTEN,:,2)})
exten = _allow_caller_id.,n,GotoIf($[${LEN(${MyCallerID})}10]?NoCID)
exten = _allow_caller_id.,n,Set(CALLERID(num)=${MyCallerID})
exten = _allow_caller_id.,n(NoCID),Set(MyNumber=${CUT(EXTEN,:,3)})
exten = _allow_caller_id.,n,Set(MyTime=${CUT(EXTEN,:,4)})
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} calling ${MyNumber} for
${MyTime} seconds)
exten = _allow_caller_id.,n,Set(CALLERPRES()=allowed_not_screened)
exten = _allow_caller_id.,n,Dial${OutBoundDev}/${MyNumber},${MyTime},g)
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} call just got status
${DIALSTATUS})
exten = _allow_caller_id.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 22, 2010, at 1:31 PM, mancyb...@gmail.com wrote:
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately the 'Variable' parameter is used to push data between the
originating script and the dialplan, not commands.
Example:
Variable: var1=23|var2=24|var3=25
Additionally, this data can be used in the dialplan only when the call gets
answered or when it fails.
I can't find a way to inject the parameter DURING (or before) the call.
Thank you very much for supporting,
Mike
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Re: [asterisk-users] Hangup after n seconds using originate ?
Here is how I do it, Mike
-- Perl Code --
my $phone_number=4918802;
my $testfile = /tmp/testin_$$.wav;
unlink $testfile;
my %resp = $astman-sendcommand( Action = 'Originate',
Channel =
DAHDI/$key/w$phone_number,
Variable = ARG1=$testfile,
Exten = 'SIP/170',
Context = 'testit',
ApplicationID = 1,
priority = 1,
Number = $phone_number
);
Context
[testit]
exten = s,1,Answer(1)
exten = s,n,Progress()
exten = s,n,SetMusicOnHold(default)
exten = s,n,Waitexten(5,m)
exten = s,n,Verbose(record ${ARG1})
exten = s,n,record(${ARG1}|0|10|s)
exten = s,n,Waitexten(5,m)
exten = s,n,Goto(end-call|s|1)
Context 2
[end-call]
exten = s,1,Verbose(details - time ${DIALEDTIME} time2 ${ANSWEREDTIME}
status ${DIALSTATUS})
exten = s,n,AGI(clearorder.agi|${ABA}|${CHANNEL(language)})
exten = s,n,GotoIf($[${HANGUPCAUSE} = 0]?end-call|h|1)
exten = s,n,playback(vm-goodbye|noanswer)
exten = h,1,Hangup(${HANGUP_CAUSE})
This snippet calls 205-491-8802 (Telco Test line) and records 10 seconds of
tone into a file, then hangs up.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
mancyb...@gmail.com
Sent: Thursday, April 22, 2010 3:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup after n seconds using originate ?
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something
like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately the 'Variable' parameter is used to push data between the
originating script and the dialplan, not commands.
Example:
Variable: var1=23|var2=24|var3=25
Additionally, this data can be used in the dialplan only when the call gets
answered or when it fails.
I can't find a way to inject the parameter DURING (or before) the call.
Thank you very much for supporting,
Mike
--
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Re: [asterisk-users] Portech MV-374 does not register behind NAT
Take out the router/firewall and connect directly to the net to test your NAT problem theory. On Thu, Apr 22, 2010 at 12:15 PM, Jonas Kellens jonas.kell...@telenet.bewrote: Jared, thank you for your answer. As I said in my previous mail, I'm using a Zyxel NBG-419 router (which normally supports VoIP and QoS). Firewall is disabled on the Zyxel. The MV-374 only accepts IP-address, not a FQDN. Will give it another try though... The answer from Portech-support : use STUN. Even if the NAT rewrites the IP-address/port combination, why is it a problem for the Portech and not for the IP-phones (Grandstream Snom) ? They all communicate on port 5060 -- 5064 (several SIP-accounts) Jonas. Jared Smith wrote: On Thu, 2010-04-22 at 17:45 +0200, Jonas Kellens wrote: All goes well when the gateway is connected directly to the internet... It's when it is behind NAT the 401 is sent from Asterisk... Is the device registering to an IP address, or do a DNS name? What type of NAT firewall are you using? This reminds me of a problem I had years ago with a Cisco PIX firewall, where it would rewrite IP addresses in the SIP Request URI, causing the authentication to fail. One solution was to have it register to a fully-qualified domain name instead of an IP address, so that the Request URI wouldn't get overwritten. It's certainly worth a shot... -- Jared Smith Digium, Inc. -- _ -- 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] More efficient dial plan for a list of selective inbound numbers
I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.) [custom-inbound] exten = _556,1,answer exten = _556,n,playback(beep) exten = _557,1,answer exten = _557,n,playback(beep) exten = _558,1,answer exten = _558,n,playback(beep) exten = _989,1,answer exten = _989,n,playback(beep) If there are like 100s of different prefixes, this list gets really big. Not desired. How can I have a more efficient dialplan? Thanks, Bruce -- _ -- 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] More efficient dial plan for a list of selectiveinbound numbers
Use an AGI to do a database lookup and return a value
[custom-inbound]
Exten = s,1,AGI(lookup.agi,${EXTEN})
Exten = s,n,GotoIf($[${AGI_RETURN} = blah}?1:2)
I would recommend the built-in database for simplicity, but for hundreds of
numbers you are going to want either a compiled C agi and/or a real
database lookup.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Thursday, April 22, 2010 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] More efficient dial plan for a list of
selectiveinbound numbers
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much longer.
List of prefixes (556,557,557,989.)
[custom-inbound]
exten = _556,1,answer
exten = _556,n,playback(beep)
exten = _557,1,answer
exten = _557,n,playback(beep)
exten = _558,1,answer
exten = _558,n,playback(beep)
exten = _989,1,answer
exten = _989,n,playback(beep)
If there are like 100s of different prefixes, this list gets really big. Not
desired. How can I have a more efficient dialplan?
Thanks,
Bruce
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Re: [asterisk-users] More efficient dial plan for a list of selective inbound numbers
Catches 555 through 559: exten = _55[5-9],1,answer exten = _55[5-9],n,playback(beep) http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns -- _ -- 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] Portech MV-374 does not register behind NAT
As I already said 2 times earlier in this thread : when I connect directly to the internet, then the registration goes through normally. So according to me it is definitely a NAT-problem. Don't need to find this out another 20 times. Just don't know just what setting is needed when the Portech is behind NAT... Jonas. bruce bruce wrote: Take out the router/firewall and connect directly to the net to test your NAT problem theory. On Thu, Apr 22, 2010 at 12:15 PM, Jonas Kellens jonas.kell...@telenet.be wrote: Jared, thank you for your answer. As I said in my previous mail, I'm using a Zyxel NBG-419 router (which normally supports VoIP and QoS). Firewall is disabled on the Zyxel. The MV-374 only accepts IP-address, not a FQDN. Will give it another try though... The answer from Portech-support : "use STUN". Even if the NAT rewrites the IP-address/port combination, why is it a problem for the Portech and not for the IP-phones (Grandstream Snom) ? They all communicate on port 5060 -- 5064 (several SIP-accounts) Jonas. -- _ -- 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 do analog em on asterisk?
Zeeshan Zakaria wrote: Can anybody with previous experience with it guide me on how to setup asterisk with analog em to connect it to an old style em system which uses 4 pair cables on RJ 45 jacks. All the analog cards I know of use RJ 11 jacks. And there is no choice of modernization of the customer equipment. To my knowledge, there are no analog cards available for DAHDI that have EM ports on them. The only way to provide analog EM support from Asterisk over DAHDI is to connect a port on a T-1 card to a channel bank with EM cards in it. These should be pretty easy to find on the used market. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] Follow-me to my answering machine :-(
Ryan, Thanks, but as I said, part of the problem is that I can't use DTMF in my car. So having to 'press 1' is unacceptable. Bryan Jacobs On Thu, 22 Apr 2010 15:54:47 -0400 Ryan Bullock rrb3...@gmail.com wrote: Check out the 'p' option for the Dial command. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial It enables call screening, so you have to press 1 to answer. This can also prevent the voice mail from being left on your cell phone. signature.asc Description: PGP signature -- _ -- 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] Follow-me to my answering machine :-(
Motiejus, I'm not sure my cell phone plays these - the behavior I observe is that the call is forwarded to an external number I can control if: a) The cell phone is out of the service area or off or b) I'm busy or reject the call Currently, I have this number set to my Asterisk direct-to-voicemail DID. Again, I *want* to leave these forwarding settings in place because they mean that calls to my cell phone DID can go to my Asterisk voicemail. Which is good. Is there some tone played before the call is forwarded? I hadn't noticed one. Bryan Jacobs On Thu, 22 Apr 2010 21:00:31 +0300 Motiejus Jakštys desired@gmail.com wrote: Hi, currently I am writing a sound recognition software that will suit here pretty well - it can recognize your cell phone's our of radio coverage or similar operator message. It's GPL, link here: http://github.com/Motiejus/SoundPatty Now the program can say if 2 WAV files match (tested with out of radio coverage status and GSM network - it works), and right now I am working with it's support with asterisk (through JACK_HOOK). It shouldn't take more than a week, I hope. I will announce to this conference when it's ready :) Regards Motiejus On Thu, Apr 22, 2010 at 8:31 PM, Bryan Jacobs n...@landwarsin.asia wrote: Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that would be bad (TM). app_followme sounds like a solution. BUT, I also have a car. And I cannot use DTMF to respond to the app_followme prompts (which I WANT, to avoid routing the forwarded call to voicemail when the cell phone is off and its voicemail picks up), while driving. I've tried using dial macros and AMD(), but this is complex, very unreliable, and delays the connection of the call significantly. Is there some way to make app_followme use voice recognition? Or some other solution so that I can get all my phones to ring with one number, even when my cell phone is off or out of range? Bryan Jacobs -- _ -- 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 signature.asc Description: PGP signature -- _ -- 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] Follow-me to my answering machine :-(
You could use the non-followme option from this link http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe and use Lumenvox or Vestec ($50 or $25 for a 1 port license) to be able to verbally do the 1/yes/2/no thing. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan Jacobs Sent: Thursday, April 22, 2010 4:37 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Follow-me to my answering machine :-( Motiejus, I'm not sure my cell phone plays these - the behavior I observe is that the call is forwarded to an external number I can control if: a) The cell phone is out of the service area or off or b) I'm busy or reject the call Currently, I have this number set to my Asterisk direct-to-voicemail DID. Again, I *want* to leave these forwarding settings in place because they mean that calls to my cell phone DID can go to my Asterisk voicemail. Which is good. Is there some tone played before the call is forwarded? I hadn't noticed one. Bryan Jacobs On Thu, 22 Apr 2010 21:00:31 +0300 Motiejus Jakštys desired@gmail.com wrote: Hi, currently I am writing a sound recognition software that will suit here pretty well - it can recognize your cell phone's our of radio coverage or similar operator message. It's GPL, link here: http://github.com/Motiejus/SoundPatty Now the program can say if 2 WAV files match (tested with out of radio coverage status and GSM network - it works), and right now I am working with it's support with asterisk (through JACK_HOOK). It shouldn't take more than a week, I hope. I will announce to this conference when it's ready :) Regards Motiejus On Thu, Apr 22, 2010 at 8:31 PM, Bryan Jacobs n...@landwarsin.asia wrote: Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that would be bad (TM). app_followme sounds like a solution. BUT, I also have a car. And I cannot use DTMF to respond to the app_followme prompts (which I WANT, to avoid routing the forwarded call to voicemail when the cell phone is off and its voicemail picks up), while driving. I've tried using dial macros and AMD(), but this is complex, very unreliable, and delays the connection of the call significantly. Is there some way to make app_followme use voice recognition? Or some other solution so that I can get all my phones to ring with one number, even when my cell phone is off or out of range? Bryan Jacobs -- _ -- 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] Follow-me to my answering machine :-(
Danny,
That sounds like a decent idea. The dial screening macros are not well
documented and difficult to get right (for example: if one channel
returns BUSY and another returns CONTINUE, what happens?).
I feel that this should be an option built into app_followme - if there
were a confirmation={none,dtmf,voice} option for each leg, this would
be much easier to implement cleanly.
Bryan Jacobs
On Thu, 22 Apr 2010 16:42:24 -0500
Danny Nicholas da...@debsinc.com wrote:
You could use the non-followme option from this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
and use Lumenvox or Vestec ($50 or $25 for a 1 port license) to be
able to verbally do the 1/yes/2/no thing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan
Jacobs Sent: Thursday, April 22, 2010 4:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Follow-me to my answering machine :-(
Motiejus,
I'm not sure my cell phone plays these - the behavior I observe is
that the call is forwarded to an external number I can control if:
a) The cell phone is out of the service area or off
or
b) I'm busy or reject the call
Currently, I have this number set to my Asterisk direct-to-voicemail
DID.
Again, I *want* to leave these forwarding settings in place because
they mean that calls to my cell phone DID can go to my Asterisk
voicemail. Which is good.
Is there some tone played before the call is forwarded? I hadn't
noticed one.
Bryan Jacobs
On Thu, 22 Apr 2010 21:00:31 +0300
Motiejus Jakštys desired@gmail.com wrote:
Hi,
currently I am writing a sound recognition software that will suit
here pretty well - it can recognize your cell phone's our of radio
coverage or similar operator message. It's GPL, link here:
http://github.com/Motiejus/SoundPatty
Now the program can say if 2 WAV files match (tested with out of
radio coverage status and GSM network - it works), and right now I
am working with it's support with asterisk (through JACK_HOOK). It
shouldn't take more than a week, I hope.
I will announce to this conference when it's ready :)
Regards
Motiejus
On Thu, Apr 22, 2010 at 8:31 PM, Bryan Jacobs n...@landwarsin.asia
wrote:
Hello asterisk users!
I, like many people, have a cell phone. I also have some SIP
phone devices (software and hardware). I'd like to have one
number that rings all my phones and routes the call to wherever I
pick up.
However, my cell phone has its own call forwarding voicemail. I
can't just turn that off, because then direct-to-cell calls
wouldn't ever get to voicemail - that would be bad (TM).
app_followme sounds like a solution. BUT, I also have a car.
And I cannot use DTMF to respond to the app_followme prompts
(which I WANT, to avoid routing the forwarded call to voicemail
when the cell phone is off and its voicemail picks up), while
driving.
I've tried using dial macros and AMD(), but this is complex,
very unreliable, and delays the connection of the call
significantly.
Is there some way to make app_followme use voice recognition? Or
some other solution so that I can get all my phones to ring with
one number, even when my cell phone is off or out of range?
Bryan Jacobs
--
_
-- 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
signature.asc
Description: PGP signature
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] Follow-me to my answering machine :-(
Maybe I'll get brave and try this as a patch :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan Jacobs
Sent: Thursday, April 22, 2010 4:57 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Follow-me to my answering machine :-(
Danny,
That sounds like a decent idea. The dial screening macros are not well
documented and difficult to get right (for example: if one channel
returns BUSY and another returns CONTINUE, what happens?).
I feel that this should be an option built into app_followme - if there
were a confirmation={none,dtmf,voice} option for each leg, this would
be much easier to implement cleanly.
Bryan Jacobs
On Thu, 22 Apr 2010 16:42:24 -0500
Danny Nicholas da...@debsinc.com wrote:
You could use the non-followme option from this link
http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
and use Lumenvox or Vestec ($50 or $25 for a 1 port license) to be
able to verbally do the 1/yes/2/no thing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan
Jacobs Sent: Thursday, April 22, 2010 4:37 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Follow-me to my answering machine :-(
Motiejus,
I'm not sure my cell phone plays these - the behavior I observe is
that the call is forwarded to an external number I can control if:
a) The cell phone is out of the service area or off
or
b) I'm busy or reject the call
Currently, I have this number set to my Asterisk direct-to-voicemail
DID.
Again, I *want* to leave these forwarding settings in place because
they mean that calls to my cell phone DID can go to my Asterisk
voicemail. Which is good.
Is there some tone played before the call is forwarded? I hadn't
noticed one.
Bryan Jacobs
On Thu, 22 Apr 2010 21:00:31 +0300
Motiejus Jakštys desired@gmail.com wrote:
Hi,
currently I am writing a sound recognition software that will suit
here pretty well - it can recognize your cell phone's our of radio
coverage or similar operator message. It's GPL, link here:
http://github.com/Motiejus/SoundPatty
Now the program can say if 2 WAV files match (tested with out of
radio coverage status and GSM network - it works), and right now I
am working with it's support with asterisk (through JACK_HOOK). It
shouldn't take more than a week, I hope.
I will announce to this conference when it's ready :)
Regards
Motiejus
On Thu, Apr 22, 2010 at 8:31 PM, Bryan Jacobs n...@landwarsin.asia
wrote:
Hello asterisk users!
I, like many people, have a cell phone. I also have some SIP
phone devices (software and hardware). I'd like to have one
number that rings all my phones and routes the call to wherever I
pick up.
However, my cell phone has its own call forwarding voicemail. I
can't just turn that off, because then direct-to-cell calls
wouldn't ever get to voicemail - that would be bad (TM).
app_followme sounds like a solution. BUT, I also have a car.
And I cannot use DTMF to respond to the app_followme prompts
(which I WANT, to avoid routing the forwarded call to voicemail
when the cell phone is off and its voicemail picks up), while
driving.
I've tried using dial macros and AMD(), but this is complex,
very unreliable, and delays the connection of the call
significantly.
Is there some way to make app_followme use voice recognition? Or
some other solution so that I can get all my phones to ring with
one number, even when my cell phone is off or out of range?
Bryan Jacobs
--
_
-- 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:
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Re: [asterisk-users] How to do analog em on asterisk?
Thanks Kevin for your reply. We tried this option with two MultiVoIP devices but results were not satisfactory. I was hoping I could do it without any external device. My team doesn't want to take any more third party-asterisk integration risk for this mission critical communication system after bad MultiVoIP experience. I know there is an Adtran device which does it all in one box, and we have it deployed at some sites and they work flawlessly, but they are expensive too. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-22 5:30 PM, Kevin P. Fleming kpflem...@digium.com wrote: Zeeshan Zakaria wrote: Can anybody with previous experience with it guide me on how to setup as... To my knowledge, there are no analog cards available for DAHDI that have EM ports on them. The only way to provide analog EM support from Asterisk over DAHDI is to connect a port on a T-1 card to a channel bank with EM cards in it. These should be pretty easy to find on the used market. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Follow-me to my answering machine :-(
If you're saying the equipment in your car won't generate DTMF tones, a quick-and-dirty solution would be to use a pocket DTMF dialer. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan Jacobs Sent: Thursday, April 22, 2010 4:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Follow-me to my answering machine :-( Ryan, Thanks, but as I said, part of the problem is that I can't use DTMF in my car. So having to 'press 1' is unacceptable. Bryan Jacobs On Thu, 22 Apr 2010 15:54:47 -0400 Ryan Bullock rrb3...@gmail.com wrote: Check out the 'p' option for the Dial command. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial It enables call screening, so you have to press 1 to answer. This can also prevent the voice mail from being left on your cell phone. -- _ -- 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] Follow-me to my answering machine :-(
Ah, sorry, I totally missed that in your description. Other than the speech recognition that Danny is suggesting, my only thought is to use an agi that will originate another leg, run AMD (answering machine detect) and then dump the two parties into a conference to re-join them(or use the Bridge command in newer version). -- _ -- 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 do analog em on asterisk?
Zeeshan Zakaria wrote: Thanks Kevin for your reply. We tried this option with two MultiVoIP devices but results were not satisfactory. I was hoping I could do it without any external device. My team doesn't want to take any more third party-asterisk integration risk for this mission critical communication system after bad MultiVoIP experience. I know there is an Adtran device which does it all in one box, and we have it deployed at some sites and they work flawlessly, but they are expensive too. MultiVOIP is not a channel bank, it's a SIP media gateway. Look for a Carrier Access ADIT 600 or something similar; pretty much any modular T1 channel bank should have EM cards available. EM will be expensive (somewhat) no matter which route you take, because it's not something that was deployed in massive volumes like FXS/FXO are, and it was primarily used for trunking between large expensive PBXes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] More efficient dial plan for a list of selective inbound numbers
If all the dialplan follow the exact same patten, you may try use realtime and put the dialplan into mysql. Just my 2 cents. bruce bruce wrote: I have a list of CLIDs prefixes that I want to use in a context. Basically, I want to do this but the list of prefix numbers is much longer. List of prefixes (556,557,557,989.) [custom-inbound] exten = _556,1,answer exten = _556,n,playback(beep) exten = _557,1,answer exten = _557,n,playback(beep) exten = _558,1,answer exten = _558,n,playback(beep) exten = _989,1,answer exten = _989,n,playback(beep) If there are like 100s of different prefixes, this list gets really big. Not desired. How can I have a more efficient dialplan? Thanks, Bruce -- Jian Gao IT Technician SJ Geophysics Ltd. http://www.sjgeophysics.com jian@sjgeophysics.com mailto:jian@sjgeophysics.com Tel: (604)582-1100 -- _ -- 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] Follow-me to my answering machine :-(
Don, No, I'm not trying to say there's a problem with generating the tones. The issue is that my phone is still holstered, connected to the car via Bluetooth. I have steering-wheel buttons for receiving calls and hanging up, but I don't have a safe way to press buttons. Bryan Jacobs On Thu, 22 Apr 2010 17:04:29 -0500 Don Kelly d...@donkelly.biz wrote: If you're saying the equipment in your car won't generate DTMF tones, a quick-and-dirty solution would be to use a pocket DTMF dialer. --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryan Jacobs Sent: Thursday, April 22, 2010 4:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Follow-me to my answering machine :-( Ryan, Thanks, but as I said, part of the problem is that I can't use DTMF in my car. So having to 'press 1' is unacceptable. Bryan Jacobs On Thu, 22 Apr 2010 15:54:47 -0400 Ryan Bullock rrb3...@gmail.com wrote: Check out the 'p' option for the Dial command. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial It enables call screening, so you have to press 1 to answer. This can also prevent the voice mail from being left on your cell phone. signature.asc Description: PGP signature -- _ -- 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 do analog em on asterisk?
Thank you for this info. I'll look into this equipment and other similar. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-22 6:27 PM, Kevin P. Fleming kpflem...@digium.com wrote: Zeeshan Zakaria wrote: Thanks Kevin for your reply. We tried this option with two MultiVoIP devi... MultiVOIP is not a channel bank, it's a SIP media gateway. Look for a Carrier Access ADIT 600 or something similar; pretty much any modular T1 channel bank should have EM cards available. EM will be expensive (somewhat) no matter which route you take, because it's not something that was deployed in massive volumes like FXS/FXO are, and it was primarily used for trunking between large expensive PBXes. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsvill... -- _ -- 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 running @ 100% load doing nothing
Hi guys, I just ran into a funny issue here. I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a quick glance of the system: * Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer) everything up2date. * Asterisk 1.6.2.6 If I run asterisk using the debian init script in contrib/init.d/, top shows asterisk is using 99.x% CPU doing nothing. If I run asterisk with -vvvc, it runs just fine. It's the same thing either running under root or asterisk (set in asterisk.conf and confirmed with top). I checked my logs, nothing weird. It's not spitting a million error messages. It's not crashing. So any hints? What can I provide you guys to help me out? Here are some log outputs: (nothing weird) - /etc/asterisk/logger.conf [logfiles] debug = debug console = notice,warning,error messages = notice,warning,error - debug -- [Apr 22 17:50:48] DEBUG[2550] xmldoc.c: Cannot find variable 'SIPPEER' in tree 'description' [Apr 22 17:50:48] DEBUG[2550] xmldoc.c: Cannot find variable 'SIPCHANINFO' in tree 'description' message - [Apr 22 17:50:48] NOTICE[2550] cdr.c: CDR simple logging enabled. [Apr 22 17:50:48] NOTICE[2550] loader.c: 33 modules will be loaded. [Apr 22 17:50:48] WARNING[2550] translate.c: plc_samples 160 format f [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '104' is now Reachable. (32ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '105' is now Reachable. (34ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '107' is now Reachable. (30ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '101' is now Reachable. (31ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '103' is now Reachable. (29ms / 2000ms) [Apr 22 17:50:48] NOTICE[2561] chan_sip.c: Peer '102' is now Reachable. (32ms / 2000ms) [Apr 22 17:50:49] NOTICE[2561] chan_sip.c: Peer '109' is now Reachable. (30ms / 2000ms) [Apr 22 17:50:49] NOTICE[2561] chan_sip.c: Peer '203' is now Reachable. (51ms / 2000ms) [Apr 22 17:50:49] NOTICE[2561] chan_sip.c: Peer '202' is now Reachable. (56ms / 2000ms) [Apr 22 17:50:49] NOTICE[2561] chan_sip.c: Peer '110' is now Reachable. (32ms / 2000ms) [Apr 22 17:50:49] NOTICE[2561] chan_sip.c: Peer '201' is now Reachable. (54ms / 2000ms) kelvin NOTICE: This communication is intended only for the use of the person or entity named above and may contain information that is confidential or legally privileged. If you are not the intended recipient named above or a person responsible for delivering messages or communication to the intended recipient, you are hereby notified that any use, distribution, or copying of this communication or any of the information contained in it is strictly prohibited. If you have this communication in error, please notify me immediately by telephone and then destroy or delete this communication, or return it to me by mail if requested. Thank you for your attention and cooperation. -- _ -- 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] Hangup after n seconds using originate ?
Thanks for the comments, this did the trick :)
On Thu, 22 Apr 2010 13:51:35 -0700
Jim Dickenson dicken...@cfmc.com wrote:
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel: Local/allow_caller_id:415111:541222:3...@context
Exten: do_echo
Context: cfmc_cdi_private
Priority: 1
Variable: CfMC_ActionID=AllowCallerID
ActionID: AllowCallerID
Async: true
exten = _allow_caller_id.,1,Verbose(1,allow_caller_id gets ${EXTEN})
exten = _allow_caller_id.,n,Set(MyCallerID=${CUT(EXTEN,:,2)})
exten = _allow_caller_id.,n,GotoIf($[${LEN(${MyCallerID})}10]?NoCID)
exten = _allow_caller_id.,n,Set(CALLERID(num)=${MyCallerID})
exten = _allow_caller_id.,n(NoCID),Set(MyNumber=${CUT(EXTEN,:,3)})
exten = _allow_caller_id.,n,Set(MyTime=${CUT(EXTEN,:,4)})
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} calling ${MyNumber} for
${MyTime} seconds)
exten = _allow_caller_id.,n,Set(CALLERPRES()=allowed_not_screened)
exten = _allow_caller_id.,n,Dial${OutBoundDev}/${MyNumber},${MyTime},g)
exten = _allow_caller_id.,n,Verbose(1,${OutBoundDev} call just got status
${DIALSTATUS})
exten = _allow_caller_id.,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Apr 22, 2010, at 1:31 PM, mancyb...@gmail.com wrote:
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately the 'Variable' parameter is used to push data between the
originating script and the dialplan, not commands.
Example:
Variable: var1=23|var2=24|var3=25
Additionally, this data can be used in the dialplan only when the call gets
answered or when it fails.
I can't find a way to inject the parameter DURING (or before) the call.
Thank you very much for supporting,
Mike
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Re: [asterisk-users] Time difference in CSV CDR's and MySQL CDR's
2010/4/22 Steve Edwards asterisk@sedwards.com: Un-top-posting... 2010/4/22 Alejandro Recarey alexreca...@gmail.com: I am having a curious problem. I use two cdr backends, csv and MySQL. I am finding that the calldate field varies between 3 seconds and 3 minutes between the MySQL database and the CSV files! Is this expected behaviour? I thought they should both use the same timestamp. On Thu, 22 Apr 2010, Zhang Shukun wrote: the time in the file cdr is right, as mysql. calldate is the time when the record insert into mysql. I'm just a 1.2 Luddite, but... In cdr_addon_mysql.c: localtime_r(cdr-start.tv_sec,tm); strftime(timestr,128,DATE_FORMAT,tm); and then timestr is used to populate the 'calldate' column when the insert statement is built. Which is consistent with my CDRs -- they show the time the call was started, not some time after the call is finished when the row is inserted into the database. but in the cdr_mysql.conf, it said as following: ; Older versions of cdr_mysql set the calldate field to whenever the ; record was posted, rather than the start date of the call. This flag ; reverts to the old (incorrect) behavior. Note that you'll also need ; to comment out the start=calldate alias, below, to use this. compat=no i use asterisk 1.6.2.1 -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks for your supporting, have a nice day. Sucan -- _ -- 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 running @ 100% load doing nothing
I just ran into a funny issue here. I'm trying to virtualize our asterisk pbx onto vmware esxi. Here's a quick glance of the system: * Ubuntu 9.10 i386 with linux-rt kernel (to get 1000Hz timer) everything up2date. * Asterisk 1.6.2.6 If I run asterisk using the debian init script in contrib/init.d/, top shows asterisk is using 99.x% CPU doing nothing. If I run asterisk with -vvvc, it runs just fine. It's the same thing either running under root or asterisk (set in asterisk.conf and confirmed with top). I checked my logs, nothing weird. It's not spitting a million error messages. It's not crashing. And I've just done another test. With stock ubuntu 9.10 i386 and sample asterisk config files, I have the same result. VMWare shows no crazy stats of disk access nor memory usage. Just 100% cpu load. I selected Mail Server and OpenSSH server at tasksel screen during installation. Same thing. So I uninstall-all and apt-get install asterisk. Same thing. 100% with init script but normal with -c CLI switch. BTW, my server has been running for 2 days without crashing @ 100% load of course. Weird, eh. kel NOTICE: This communication is intended only for the use of the person or entity named above and may contain information that is confidential or legally privileged. If you are not the intended recipient named above or a person responsible for delivering messages or communication to the intended recipient, you are hereby notified that any use, distribution, or copying of this communication or any of the information contained in it is strictly prohibited. If you have this communication in error, please notify me immediately by telephone and then destroy or delete this communication, or return it to me by mail if requested. Thank you for your attention and cooperation. -- _ -- 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] High Availability - Shared Database - Ideas?
On Wed, Apr 21, 2010 at 1:09 PM, Robert Grignon rgrig...@fleetone.com wrote: I am investigating High Availability solutions for my front end servers. Always good to hear. I got into a discussion regarding replicated local databases versus a single fiber connected shared database on an EMC. I will guess that you mean MySQL Master/Slave replication. Is anyone running a shared database on a SAN? Care to comment on your findings... I am running MySQL on shared SAN LUN, but not for Asterisk. Since SANs are expensive, I have been using DRBD/GFS2/MySQL for most of my low budget HA Asterisk installations. Some things to think about: 1. If you are using MySQL, then only one server can have the database open at a time. You will have some lag/downtime when the active server fails and the secondary has to take over. You are going to have this anyway even with a Master/Master replication as the IP has to shift. Same with Master/Slave plus you add time for a script to promote the Slave. 2. Don't even think about using MyISAM... InnoDB *only*. MyISAM doesn't check improperly closed tables until they are accessed which can cause some major lag. Not to mention no transaction support. You won't have another copy if things get corrupted (besides all of your backups of course) 3. While nice SANs are redundant, you are still adding another dependency to the system (a few if you are using FC switches). Make sure everything has multiple paths, and don't forget to configure fencing for the nodes. 4. If you have PRI/Analog lines to the server, then it becomes more of a headache. Use dependable redundant SIP gateways, or have some action plan in place. 5. Test, test, test then test some more. Break it in the lab and know how to fix it. Setup is easy, repair can be a pain. (You also want to know it will actually work =) That's my quick $0.02, and there is a lot more to think about too. Overall, if designed right I think it is a good option. Just depends on your level of comfort with the technologies, and the risk/benefit that goes along with it. -Jonathan -- _ -- 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] Hans Rauser
http://shotojukuindia.com/default/index.php -- _ -- 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] More efficient dial plan for a list of selective inbound numbers
bruce bruce wrote:
I have a list of CLIDs prefixes that I want to use in a context.
Basically, I want to do this but the list of prefix numbers is much
longer. List of prefixes (556,557,557,989.)
[custom-inbound]
exten = _556,1,answer
exten = _556,n,playback(beep)
exten = _557,1,answer
exten = _557,n,playback(beep)
exten = _558,1,answer
exten = _558,n,playback(beep)
exten = _989,1,answer
exten = _989,n,playback(beep)
If there are like 100s of different prefixes, this list gets really big.
Not desired. How can I have a more efficient dialplan?
You could use a pattern match and then do a lookup in the database to see if
the
prefix exists:
[custom-inbound]
exten = _XXX,1,Verbose(2,Lookup prefix ${EXTEN})
exten = _XXX,n,Set(PREFIX=${EXTEN})
exten = _XXX,n,Set(PREFIX_EXISTS=${ODBC_GET_PREFIX(${PREFIX})})
exten = _XXX,n,GotoIf($[${PREFIX_EXISTS} =
1]?prefixAllowed,1:prefixDisallowed)
exten = prefixAllowed,1,Verbose(2,Prefix ${PREFIX} Allowed)
exten = prefixAllowed,n,Playback(beep)
exten = prefixDisallowed,1,Verbose(2,Prefix ${PREFIX} Disallowed)
exten = prefixDisallowed,n,Congestion()
func_odbc.conf
[GET_PREFIX]
dsn=something setup in res_odbc.conf)
readsql=SELECT 1 FROM prefix_table WHERE prefix = '${ARG1}'
Or something like that (untested).
Leif.
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Re: [asterisk-users] asterisk running @ 100% load doing nothing
And I've just done another test. With stock ubuntu 9.10 i386 and sample asterisk config files, I have the same result. VMWare shows no crazy stats of disk access nor memory usage. Just 100% cpu load. I selected Mail Server and OpenSSH server at tasksel screen during installation. Same thing. So I uninstall-all and apt-get install asterisk. Same thing. 100% with init script but normal with -c CLI switch. Please disregard the last message. I worked on the wrong server. Thank god I have snapshots. The init script from ubuntu asterisk package resolved the problem. All I did was apt-get install asterisk on a different machine and copy that init script over. It did the trick. But it's interesting that init script shipped with ubuntu 1.6.2.6 and 1.6.2.7-rc2 both caused the same problem. Cheers, kel NOTICE: This communication is intended only for the use of the person or entity named above and may contain information that is confidential or legally privileged. If you are not the intended recipient named above or a person responsible for delivering messages or communication to the intended recipient, you are hereby notified that any use, distribution, or copying of this communication or any of the information contained in it is strictly prohibited. If you have this communication in error, please notify me immediately by telephone and then destroy or delete this communication, or return it to me by mail if requested. Thank you for your attention and cooperation. -- _ -- 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 running @ 100% load doing nothing
I opened a ticket about this: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17217 Remove -c on the init script of asterisk, line 85. Should help. I was trying it with a xen guest. On Fri, Apr 23, 2010 at 6:52 AM, Kelvin Chan kelvin.c...@positronics.com wrote: And I've just done another test. With stock ubuntu 9.10 i386 and sample asterisk config files, I have the same result. VMWare shows no crazy stats of disk access nor memory usage. Just 100% cpu load. I selected Mail Server and OpenSSH server at tasksel screen during installation. Same thing. So I uninstall-all and apt-get install asterisk. Same thing. 100% with init script but normal with -c CLI switch. Please disregard the last message. I worked on the wrong server. Thank god I have snapshots. The init script from ubuntu asterisk package resolved the problem. All I did was apt-get install asterisk on a different machine and copy that init script over. It did the trick. But it's interesting that init script shipped with ubuntu 1.6.2.6 and 1.6.2.7-rc2 both caused the same problem. Cheers, kel NOTICE: This communication is intended only for the use of the person or entity named above and may contain information that is confidential or legally privileged. If you are not the intended recipient named above or a person responsible for delivering messages or communication to the intended recipient, you are hereby notified that any use, distribution, or copying of this communication or any of the information contained in it is strictly prohibited. If you have this communication in error, please notify me immediately by telephone and then destroy or delete this communication, or return it to me by mail if requested. Thank you for your attention and cooperation. -- _ -- 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
