Re: [asterisk-users] Asterisk Web Meetme module not loading
Matt Riddell lists at venturevoip.com writes: On 1/09/09 5:19 PM, Glen Ganderton wrote: app_cbmysql.c:37:1: warning: AST_MODULE redefined command-line: warning: this is the location of the previous definition app_cbmysql.c: In function âcheckMaxâ: app_cbmysql.c:116: warning: implicit declaration of function âast_say_numberâ app_cbmysql.c: In function âroomQueryâ: app_cbmysql.c:181: warning: unused variable âeatimeâ app_cbmysql.c:337: warning: control reaches end of non-void function I'm not sure how Asterisk is supposed to know that this requires a link to MySQL without being told. Are you using the latest version of the app_cbmysql? It looks like it needs to be updated for the latest version. Alternatively it may say somewhere on their website which version of Asterisk this works with? Dear Mr. Riddell, Greetings!!! trust this reaches you to the best of everything.. I have been encountering error whenever i run make install to load cbmysql. Below is the error. app_cbmysql.c:529:38: error: macro ast_config_load requires 2 arguments, but only 1 given app_cbmysql.c: In function âload_configâ: How i can resolve this problem. Please help. -- _ -- 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] Call Transfer question
Hi, is possible that two sip extensions: user-1 and user-2 are connected and I want to transfer the call from user-1 to a third user user-3. I know it is possible through feature keys mapping in features.conf, but I want to do this through AMI or Asterisk CLI Commands? Please suggest if possible? Thank you! Muhammad Faheem -- _ -- 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] Call Transfer question
Hi faheem, You can do this: ACTION: Redirect Channel: Channel ID Context: Context Exten: Exten Priority: Priority Regards, Qasim On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote: Hi, is possible that two sip extensions: user-1 and user-2 are connected and I want to transfer the call from user-1 to a third user user-3. I know it is possible through feature keys mapping in features.conf, but I want to do this through AMI or Asterisk CLI Commands? Please suggest if possible? Thank you! Muhammad Faheem -- _ -- 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] AstriCon 2013 (our 10th AstriCon) needs YOU!
As we plan for our 10th AstriCon, which will be in Atlanta, GA the week commencing October 7th, we want to make sure that our conference sessions are the best they've ever been! That's why we need YOU to submit a speaking proposal - to share you experiences and ideas around Asterisk! The best sessions at AstriCon are always those from the Community that show what you've been up to with Asterisk, new ways that you've put Asterisk to work, etc. - where you're showing a use case or some additional stuff that you've developed within, or in addition to, Asterisk - we're waiting to hear from you... If your speaking submission is accepted, you will get a free all-access pass pass for the whole event - worth $495 (and that's the early bird price, lasts until the end of June). Please make your speaking submission at https://docs.google.com/forms/d/1Li-hRCtqOQHpl6RHMt0PX6SvWGbwqPfKsOfRm35Vkz8/viewform If you are a developer, do remember our AstriDevCon get-together which is traditionally on the Monday - this is where the Community discusses new features and the priorities are set for future Asterisk releases. At AstriCon, we also seek to recognise those in the Community that have made great contributions - if you have any nominations for this year, please let me know. REMEMBER: If you're serious about Asterisk, you'll be at AstriCon! Registration is at http://www.asterisk.org/community/astricon-user-conference/register All the best, David Digium logo David Duffett Digium, Inc. · Director, Worldwide Asterisk Community 6 Landscape Close, Weston on the Green · Bicester, Oxfordshire OX25 3SX · UK direct/fax: +1 256 428 6119 · mobile: +44 7722 442236 twitter: dduffett · linkedin: www.linkedin.com/in/davidduffett Check us out at: http://digium.com · http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 11.4: motif can only handle one channel at a time?
I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean -- _ -- 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] 11.4: motif can only handle one channel at a time?
sean darcy wrote: I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? Motif itself has no imposed limitations, but that's not to say Google Voice doesn't. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 11.4: motif can only handle one channel at a time?
On 05/16/2013 09:41 AM, sean darcy wrote: I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean More: Two different motif sections. Two different xmpp sections. xmpp shows both connections. sean -- _ -- 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] dial and bridge
Thanks all for your help, in the end I was able to do something like: Action: Originate Channel: Local/300@from-internal/n Application: MusicOnHold Async: 1 As soon as this connects, the callee hears MOH. I get the channel out via AMI events and start another call: Action: Originate Channel: Local/301@from-internal/n Application: Bridge Data: Local/300@from-internal-aa8c;1 Async: 1 when this connects, it is immediately bridged to the first callee. I just have to keep track of errors and hang up the first call if the seconds does not go through. Thanks a lot! l. 2013/5/15 Dan Cropp d...@amtelco.com You could use AsyncAGI to achieve this. ** ** Originate the first call (passing in some unique identifier as a variable), then using AMI you will see the channel data. When you see an Event: AysncAGI for that channel (with that id, you have control of the call). Send a Dial Action telling it to dial the call and bridge them together if the person answers. If they don’t answer, you will be notified and can do something with the original call (play a message, hangup, etc). If they are bridged, you can see how long, etc. ** ** Setup an extension, naming it something like patching ** ** exten = patching,1,AGI(agi:async) ** ** Action: Originate Channel: Local/300@from-internal Async: 1 Exten: 1 Context: patching Data: 1973 Variable: YourUniquePatchID=1234 ** ** ** ** Using AsyncAGI and AMI, you can have full control of the call. You do have to setup a very simple dial plan so Asterisk knows you are using AsyncAGI to control the call. ** ** Have a great day! Dan ** ** ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Lenz Emilitri *Sent:* Tuesday, May 14, 2013 11:16 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] dial and bridge ** ** Hi all, I need some advice - I have been working on originating multiple calls using AMI and then joining them. What I want to do is: - dial call 1 (where the caller is in a channel format, like SIp/1234 or Local/1234@ext) and park it somehow - dial call 2 (where again the caller is in channel format) and join it to the previous call. ** ** As a requirement, I cannot use the dialplan as an end-point (as I cannot change it) but need to use the AMI only. ** ** I tried doing something like: ** ** Action: Originate Channel: Local/300@from-internal Async: 1 Application: Wait Data: 1973 ** ** So that the call goes to 300 and then basically stays there forever, and then I dial again: ** ** Action: Originate Channel: Local/500@from-internal Async: 1 Application: Wait Data: 1973 And then try to bridge the results, but it does not seem to work. What I would like to do would be more on the lines of: ** ** Originate call 1 and park it (using a park or waiting) Originate call 2 and bridge it immediately to call1 (using the Application part) ** ** But maybe I am missing something? is there anybody who has better suggestions? ** ** Thanks l. ** ** ** ** ** ** ** ** ** ** ** ** -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com Test-drive WombatDialer beta @ http://wombatdialer.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk High-availability/failover solutions
Hello all. As part of a project I'm working on to migrate from Asterisk 1.0.11.1 to the latest LTS version, I'm looking into providing a HA/failover solution for the new Asterisk installation I'll be deploying. It would appear my best bet would be to use the R850 Digium appliance. Does anyone have any experience running one of these devices along with AsteriskNOW? I ask because apparently from what I gather from the rseries admin manual, it's required to use flat files vs. SQL -- and obviously AsteriskNOW uses SQL. Failing that, I wonder if AsteriskNOW is nothing more than Asterisk 11.3.0 + FreePBX? Would I be fine installing 11.3+FreePBX? I'm mainly wondering what the differences would be from AsteriskNOW vs. a vanilla install. I'm currently trying to see if I can find any difference using virtual machines, but any info would be extremely helpful and time-saving :) Thanks! -- Andre Goree -=-=-=-=-=- Email - an...@drenet.net Website - http://blog.drenet.net PGP key - http://www.drenet.net/0x83ADAAAB.asc -=-=-=-=-=- -- _ -- 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 Web Meetme module not loading
Rohit Mahajan wrote: Matt Riddell lists at venturevoip.com writes: Are you using the latest version of the app_cbmysql? It looks like it needs to be updated for the latest version. Alternatively it may say somewhere on their website which version of Asterisk this works with? I have been encountering error whenever i run make install to load cbmysql. Below is the error. app_cbmysql.c:529:38: error: macro ast_config_load requires 2 arguments, but only 1 given app_cbmysql.c: In function âload_configâ: How i can resolve this problem. The best way, as the author of app_cbmysql, is to not use app_cbmysql. If you are running Asterisk 1.6.7 or later and and Web-MeetMe 4.X you can use the realtime functionality in app_meetme. The ODBC and realtime setup is a bit more complicated than app_cbmysql, but the reliability will be much better, and you won't need the equally hacky cbend.php script to handle CDR or conference shutdown events. Dan -- _ -- 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-users Digest, Vol 106, Issue 23
Hi, I am looking for the easiest and fastest way to send or pull callerID and extension# from asterisk to a web server for sales data lookup and display. User would be logged in with known extension. I have been looking at several options but was hoping someone here would have the best one. I was looking at using Master.csv but would prefer to avoid parsing it. Any suggestions much appreciated. Thanks, Nick -- _ -- 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] Planned maintenance for community services on May 16, 2013
On Thursday, May 16, 2013 , the Asterisk community services listed below will be unavailable due to maintenance being performed. This maintenance will begin at approximately 9:00 PM CDT (02:00 May 17 UTC) and should last no longer than 1 hour. The affected services are: * issues.asterisk.org * wiki.asterisk.org * code.asterisk.org * crowd.asterisk.org * signup.asterisk.org * downloads.asterisk.org * downloads.digium.com * packages.asterisk.org * asterisknow.asterisk.org Thank you for your support! -- Digium's Asterisk Development Team -- _ -- 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] Initial REGISTER Request: Contains Credentials before 401: KDDI Japan
Brian,
KDDI does provide a list of supported equipment and vendors. Specific
hardware or license based software products that quickly become cost
prohibitive.
I doubt that Asterisk will find it's way on the list any time soon. Because
KDDI follows the traditional big telco method of interoperability, which
normally means licensing products for use on their network.
That's exactly the sort of thing that the industry should be evolving away from.
For now, this should help quiet KDDI's complaints.
Here is a small patch to the transmit_register() function in chan_sip.c that
prevents adding an Authorization header to the initial REGISTER request:
--- certified-asterisk-1.8.15-cert1/channels/chan_sip.c.orig2013-05-16
16:30:12.0 -0400
+++ certified-asterisk-1.8.15-cert1/channels/chan_sip.c 2013-05-16
16:57:49.0 -0400
@@ -13620,6 +13620,8 @@
if (!ast_strlen_zero(global_useragent))
add_header(req, User-Agent, global_useragent);
+/* Never add auth header to the initial REGISTER request */
+if (r-regattempts) {
if (auth) { /* Add auth header */
add_header(req, authheader, auth);
} else if (!ast_strlen_zero(r-nonce)) {
@@ -13647,6 +13649,7 @@
ast_log(LOG_NOTICE, No authorization available for
authentication of registration to %s@%s\n, r-username, r-hostname);
}
}
+}
snprintf(tmp, sizeof(tmp), %d, r-expiry);
add_header(req, Expires, tmp);
I tested it to make sure it works as advertised, but not thoroughly enough to
be completely confident that there are no side effects. At the very least, it
should be a good starting point for a more robust patch.
Here is a SIP trace of the REGISTER dialog when Asterisk starts or when 'sip
reload' is entered at the CLI.
17:03:39.279847 IP astdev.imminc.com.sip ekiga.net.sip: SIP, length: 381
REGISTER sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0f4587f3;rport
Max-Forwards: 70
From: sip:regt...@ekiga.net;tag=as1f2818c1
To: sip:regt...@ekiga.net
Call-ID: 5f37c0dc188cdd5c02a9a092148ef217@192.168.1.1
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.15-cert1
Expires: 120
Contact: sip:s@192.168.1.1:5060
Content-Length: 0
17:03:39.370575 IP ekiga.net.sip astdev.imminc.com.sip: SIP, length: 462
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK0f4587f3;rport=42521
From: sip:regt...@ekiga.net;tag=as1f2818c1
To: sip:regt...@ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.9b08
Call-ID: 5f37c0dc188cdd5c02a9a092148ef217@192.168.1.1
CSeq: 104 REGISTER
WWW-Authenticate: Digest realm=ekiga.net,
nonce=519549c961b09f86679289f055e42960ed06592c052d
Server: Kamailio (1.5.3-notls (i386/linux))
Content-Length: 0
17:03:39.403113 IP astdev.imminc.com.sip ekiga.net.sip: SIP, length: 582
REGISTER sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK32be996c;rport
Max-Forwards: 70
From: sip:regt...@ekiga.net;tag=as36aa89ae
To: sip:regt...@ekiga.net
Call-ID: 5f37c0dc188cdd5c02a9a092148ef217@192.168.1.1
CSeq: 105 REGISTER
User-Agent: Asterisk PBX 1.8.15-cert1
Authorization: Digest username=regtest, realm=ekiga.net, algorithm=MD5,
uri=sip:ekiga.net, nonce=519549c961b09f86679289f055e42960ed06592c052d,
response=e477ad0f835211b06d750a8c3edf88ea
Expires: 120
Contact: sip:s@192.168.1.1:5060
Content-Length: 0
17:03:39.495126 IP ekiga.net.sip astdev.imminc.com.sip: SIP, length: 399
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK32be996c;rport=42521
From: sip:regt...@ekiga.net;tag=as36aa89ae
To: sip:regt...@ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.bf7b
Call-ID: 5f37c0dc188cdd5c02a9a092148ef217@192.168.1.1
CSeq: 105 REGISTER
Contact: sip:s@192.168.1.1:5060;expires=600
Server: Kamailio (1.5.3-notls (i386/linux))
Content-Length: 0
I hope this helps you convince KDDI that Asterisk is a legitimate VoIP platform.
Please reply and let me know how it goes.
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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[asterisk-users] wanpipe and digium, oslec and hardware echo canceller
Hello All; Wanpipe is working only with sangoma cards so it does not work with digium cards? Also, who is better: to have echo canceler built in with the hardware or using olsec? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
