[asterisk-users] Outgoing call queues
Hello, Is there any way to make Asterisk not drop calls that are overwhelming to the outbound trunk capacity and have it queue that call till the time the trunk is free to take a new call? 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
[asterisk-users] AMR installation error
make gives this: codec_amr.c: In function 'amrtolin_sample': codec_amr.c:227: error: 'AST_FORMAT_AMRNB' undeclared (first use in this function) codec_amr.c:227: error: (Each undeclared identifier is reported only once codec_amr.c:227: error: for each function it appears in.) codec_amr.c: In function 'lintoamr_frameout': codec_amr.c:345: warning: unused variable 'byte_count' codec_amr.c: At top level: codec_amr.c:409: error: 'AST_FORMAT_AMRNB' undeclared here (not in a function) make[1]: *** [codec_amr.o] Error 1 make: *** [codecs] Error 2 Any ideas how to fix it? -- _ -- 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] SIP Q.850 Cause
Hello, I'm trying to fetch outbound SIP PROGRESS Reason cause code in the dialplan, Asterisk 1.8.26.1 sip show settings: Q.850 Reason header:Yes Store SIP_CAUSE:Yes However, i'm not getting any value in the dialplan variables, any successful users of this feature? -- _ -- 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 support for h.324m
Hello, If anyone has successfully compiled asterisk with: app_rtsp codec_amr mp4_play mp4_save app_transcode h324m_call Please share the versions of OS software, and libraries used. Lets make this thread useful so that all tried and tested video resources of asterisk can be found in one place for ease of access and later reference. 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
[asterisk-users] AMI Originate CDRs
Hello, There seems to be a problem with asterisk cdrs when calls are generated via AMI Originate using Local channels. Asterisk writes CDR as soon as A party off-hooks. Resulting in very inaccurate billsec and duration values. Expected CDR in case of local channel origination should be 2 records, one can be when A party answers, and the other should be written when B party hangs up the phone. Any hints? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.6
Use allowguest=no And define ACLs for every SIP account. And obviously, fail2ban for blocking suspicious IPs. -- _ -- 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] meetme list concise
Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) 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] meetme and dtmf
I don't get what the 'F' option is for. Its not proper to exit a context and then reenter the conference as admin Isn't there any other way to do actions such as kick/mute/unmute users by admin dtmf trigger? On Fri, Jun 1, 2012 at 3:47 AM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 31 May 2012, Daniel Knoll wrote: is it possible to read the DTMF tones from a caller while he is in a meetme conference? I would like to read the pressed key sequence and call a command like MeetMeAdmin or System Commands. I'm using Asterisk 1.8.7. I'm just a 1.2 Luddite, but... You can use the meetme() 'X' option to jump out of the meetme and into another context. I use this to allow conference administrators to mute, un-mute, or kick users. The first digit jumps out of the meetme and into another context where I read additional digits (the user index) and then call an AGI (meetmeadmin-by-index) before returning the admin to the conference. -- 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-**usershttp://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] meetme list concise
Thanks Dan, I found the list arguments from app_meetme.c for asterisk 1.6.x There doesn't seem to be any interface for [8] = Requests Floor. How can we put initially muted users in the request to talk queue? The provision of this parameter in the meet-me source indicates this is doable... but I am unable to find an appropriate way to do it. Any hints would be great help. On Thu, Aug 15, 2013 at 11:03 PM, Dan Austin dan_aus...@phoenix.com wrote: This list was accurate up to and including Asterisk 11 ** ** [0] = Caller # [1] = Callerid Number [2] = Callerid Name [3] = Channel: [4] = 1 for Admin, NULL for User [5] = 1 for Monitor, Null otherwise [6] = 1 for Muted, NULL for UnMuted [7] = 1 for Resquests Floor, 0 otherwise [8] = 1 for 'Is Talking', 0 otherwise [9] = Call duration ** ** Dan ** ** ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *[Digital^Dude] ® *Sent:* Thursday, August 15, 2013 4:52 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] meetme list concise ** ** Hello, Can anyone tell me the format for meetme list concise command, so that I know what field is what (separated by '!'s) 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 -- _ -- 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] stop log/debug messages into /var/log/messages
Asterisk 1.6.2.10 Channel Type StatusConfiguration --- --- Console Enabled- Warning Notice Error the init script doesn't pipe in syslog/messages the /var/log/messages contains only AGI messages. On Thu, Dec 27, 2012 at 10:48 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 27 Dec 2012, [Digital^Dude] ® wrote: I disabled all logger channels but still it logs to /var/log/messages. Any hints? What version of Asterisk? What does 'logger show channels' show? Any chance your startup script pipes output through the logger shell command? -- 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 -- _ -- 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] stop log/debug messages into /var/log/messages
I disabled all logger channels but still it logs to /var/log/messages. Any hints? -- _ -- 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] Zombie Channels
Hello, I am using asterisk 1.6.x and 1.8.18.0 (LTS) on a CentOS 5 boxes. I'm using php AGI for incoming calls and after an hour of running, there is a mismatch b/w active channels (ss7) and active calls in core show channels count. Active calls are much higher than even the physical SS7 channel limit Any suggestions how I can see which channels are zombie, and how to hangup/kill them so that legitimate new calls can come in? -- 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
[asterisk-users] 16kHz sampling
Hi all, Can asterisk 1.8.x give me MixMonitor recordings of 16Khz sampling rate? Any help would be appreciated. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_ss7 quick patch to enable RBT
Hello everyone, I am trying to apply thishttp://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diffpatch on chan_ss7-2.1.0 for RingBack tone but its not accepting and throwing errors: Hunk #1 FAILED at 704. Hunk #2 FAILED at 715. I have done the patch modifications manually in l4isup.c There is just one question, how do I pass the RB file-to-play on an SS7 channel via asterisk? -- Thanks. P.S. here is the source of the patch: http://www.voip-info.org/wiki/view/chan_ss7+quick+patch+to+enable+RBT -- _ -- 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] Regrading Speech Recognition.
Is there a tool integrated with asterisk which can give us the pitch of the utterance? On Thu, Jul 12, 2012 at 3:09 PM, Satish Barot satish4aster...@gmail.comwrote: Hi Akhilesh, Probably this link would give you some idea on ASR. With the help of it, add some logic in dialplan to develop an application of your choice. (Courtesy Lefteris Zafiris) Goto https://github.com/zaf/asterisk-speech-recog/ and read README --Satish Barot On Thu, Jul 5, 2012 at 12:46 PM, akhilesh chand omakhileshch...@gmail.com wrote: ok,how can i develop with short vocab like sales,support,etc. I have read many article but I'm not able to pick the right point, how can i develop or configure speech reorganization with asterisk. Is there any article or link please share and guide me. Regards Akhilesh On Thu, Jul 5, 2012 at 12:29 PM, Mitul Limbani mi...@enterux.in wrote: Things that look simple r quite complex to build :-) Indian Accent ASR on proper names is herculean task. No speech recognition known to mankind as of date can handle so many dialects being spoken in India, so in short what you want is nice to have, but nearly impossible to develop. Better try with short vocab on generic words (sales, support, etc.) Mitul On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks Akhilesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with ss7 voice broadcast
Asterisk 1.8.7.1 built by root on a x86_64 running Linux. CentOS release 5.5 (Final) RAM: 4 GB CPU: Dual Xeon 2.66 GHz Asterisk is running as root data seg size (kbytes, -d) unlimited file size (blocks, -f) unlimited pending signals (-i) 38912 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 4096 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 38912 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited The changes in ulimit apparently don't get reflected when I run a broadcast on asterisk. -- _ -- 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 with ss7 voice broadcast
Machine specs: CentOS release 5.5 (Final) RAM: 4 GB CPU: Dual Xeon 2.66 GHz Asterisk 1.8.7.1 built by root on a x86_64 running Linux. *CLI ulimit core Core file size (core) is effectively unlimited. *CLI ulimit data Program data segment (data) is effectively unlimited. *CLI ulimit descriptors *Number of file descriptors (descriptors) is limited to 178414.* *CLI ulimit file File size (file) is effectively unlimited. *CLI ulimit locked *Amount of memory locked into RAM (locked) is limited to 32768.* *CLI ulimit memory Resident memory (memory) is effectively unlimited. *CLI ulimit processes *Number of processes (processes) is limited to 38912.* *CLI ulimit stack Program stack size (stack) is effectively unlimited. *CLI ulimit time Cpu time (time) is effectively unlimited. *CLI ulimit virtual Virtual memory (virtual) is effectively unlimited. -- _ -- 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] Local Channel Resource Limit
Hello, How can I set a hard limit to the number of Local channels asterisk can spawn? -- 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
[asterisk-users] asterisk with ss7 voice broadcast
Hello, Asterisk under 90% load of SS7 calls can only withstand the voice broadcasting for 30 minutes. After around 30 minutes, it stops receiving any call hits via AMI. No errors are reported. Giving it a minute's rest makes it work for another 30 minutes. Can anyone hint to what may be causing this? -- 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
[asterisk-users] cdr_adaptive_odbc
Hello all, I have all configurations done and cdr_odbc works fine. However, cdr_adaptive_odbc doesn't work and I get the following error: relocation error: /usr/lib/libmyodbc3-3.51.26.so: symbol strmov, version libmysqlclient_15 not defined in file libmysqlclient.so.15 with link time reference Please suggest any hints as to how I can rectify the issue. 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] CDRs on multiple servers.
cdr_odbc works fine but I have trouble using cdr_adaptive_odbc. asterisk fails to start when I uncomment 'first' and 'second' groups. Here is the error: WARNING[22722] cdr_adaptive_odbc.c: No such connection 'MySQL1' in the 'first' section of cdr_adaptive_odbc.conf. Check res_odbc.conf. WARNING[22722] cdr_adaptive_odbc.c: No such connection 'MySQL2' in the 'second' section of cdr_adaptive_odbc.conf. Check res_odbc.conf. and WARNING[22823] loader.c: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache On Wed, Jun 6, 2012 at 8:34 PM, Benny Amorsen benny+use...@amorsen.dkwrote: Owais Ahmad millennium@gmail.com writes: Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. cdr_adaptive_odbc handles multiple servers. Just define several with [foo] and [bar] and it Just Works. /Benny -- _ -- 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 Atxfer
Hello, I need to return to the original call leg that I wanted to transfer the call to. in case the destination IVR has put me in a rather long queue. Please suggest a way I can hang up the atxfer leg and return to the first call leg. The hangup parameter in dial app using '*' key works only till the destination is ringing... Please share possible ways I could go about this problem. -- 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] Disable All Asterisk Features (blind xfer, disconnect, etc)
You can unload the features module maybe On Wed, May 23, 2012 at 9:18 PM, Eduardo Pimenta e...@akivasoftware.com.brwrote: Hi Guys, is there any way to disable all Asterisk Features? We are having false dtmf detections and randon calls being put on-hold and suspect that dtmf features is the cause. Changing features.conf aparently keeps the default options. Since we dont use it, is there any way to disable it? Thanks, Eduardo -- _ -- 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] Call Recording Stream
Hello, I am able to get the call recording file path of each call in the CDR. How can I get the realtime call recording streaming? -- _ -- 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] Transfer CDRs
Please share if anyone has encountered this cdr issue with call transfer. On Fri, May 18, 2012 at 5:32 PM, [Digital^Dude] ® millennium@gmail.comwrote: Hello, I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine. Each CDR entry of calls that are transferred is repeated once. Every field including uniqueid, calldate, billsec, duration, src, dst, channel, dstchannel is exactly the same. Besides adding a constraint in the database table, isn't there any way I can resolve this call transfer cdr duplication issue in asterisk csv cdrs? -- _ -- 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] Transfer CDRs
Hello, I'm using attended call transfer in asterisk 1.8.11.0 on a CentOS machine. Each CDR entry of calls that are transferred is repeated once. Every field including uniqueid, calldate, billsec, duration, src, dst, channel, dstchannel is exactly the same. Besides adding a constraint in the database table, isn't there any way I can resolve this call transfer cdr duplication issue in asterisk csv cdrs? -- _ -- 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 CDRs
I am using AMI Originate on asterisk 1.8.11 on SIP channels. I have set unanswered=yes in cdr.conf because I want to log NO ANSWER and BUSY calls. The issue is, that if a SIP peer is not registered, and an originate request is made for that peer, a null cdr entry is made as follows: ,2012-04-05 09:28:53,,2012-04-05 09:28:53,0,0,FAILED,DOCUMENTATION,, How can I fix it? Or, how can I set cdr not to log entries if a channel doesn't exist. -- _ -- 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] Bridging an Answered call in Asterisk with another call
Be it meetme or confbridge, asterisk 1.2.x or Asterisk 10.x. Is it possible
to query a channel and get its conference number in return?
On Thu, Mar 22, 2012 at 11:09 AM, Satish Barot satish4aster...@gmail.comwrote:
Jayesh, Personally I haven't worked on Congbridge :).
Confbridge has evolved a lot in 10.X. So probably you should have no
issues using it.
On Thu, Mar 22, 2012 at 11:04 AM, Jayesh Nambiar jayesh.v...@gmail.comwrote:
Thank you Satish. I was also thinking on similar lines. I was just
wondering if there was any mechanism with which we can bridge a new call
with the existing running call if the Call-ID of the call is known !!
I can definitely use the confbridge application for the same right; as I
am working on Asterisk10. What do you suggest??
Thanks again,
--- Jayesh
On Thu, Mar 22, 2012 at 10:33 AM, Satish Barot satish4aster...@gmail.com
wrote:
Make your user wait in a *Meetme* and then call your destination number
through AMI and once he answers, place him in the same *Meetme*.
e.g. Assuming your destination is SIP extension, have something like...
Action: Originate
Channel: SIP/{your_destination_here}
Application: MeetMe
Data: {your_meetme_number_here}
Hope this helps.
--Satish Barot
On Wed, Mar 21, 2012 at 9:06 PM, Jayesh Nambiar
jayesh.v...@gmail.comwrote:
Hello All,
I need to know a way of connecting an Answered call in Asterisk to
another call which was triggered by an AMI. I have a scenario as follows:
1) User dials 123 from a touch screen Polycom phone.
2) Call comes to Asterisk and Asterisk answers the call and asks for
PIN number.
3) Once the PIN is validated, Asterisk sends a User Event through AMI
which invokes a browser in the Polycom phone.
4) The Browser will have a Text-Box to Enter the destination number
where the caller wants to be bridged.
5) The caller enters this number in the browser which is sent as a
Originate command to Asterisk through the AMI. Please note Asterisk does
not get this number as DTMF events !!
6) Now, I need to BRIDGE this originated call from the AMI with the
actual caller who is already present in Answered state in Asterisk probably
listening to some music.
Is there any straightforward application or function to achieve this in
Asterisk.
Any ideas or directions will be of great help !!
Thanks,
--- Jayesh
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[asterisk-users] Processed Call Counter
Hi there How can I reset the value of asterisk' calls processed without restarting asterisk? Where does it save/access the value of all processed calls since last restart from? -- _ -- 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 proces memory increase
AFAIK: Linux has a tendency to keep RAM filled up with any recently accessed progarm. To keep programs access fast enough, it never removes something from the memory, only replaces it, in case some program has more frequent access than the one already present in ram. If your server isn't swapping.. things are okay. On Thu, Mar 8, 2012 at 1:15 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** Hello, I notice that at the end of the day, after about 4000 calls have passed my Asterisk-system, that the use of memory is very high and stays that way untill a restart of Asterisk or a reboot of the server. This is the situation at the end of the day : [root@sp asterisk]# free -m total used free sharedbuffers cached Mem: 3923 1931 1992 0268 1308 -/+ buffers/cache:353 3569 Swap: 4031 0 4031 So total of 3923 MB, where 1992 MB is used and 1931 MB still free. This memory never decreases and increases with every call that passes my Asterisk system. After some more calls : [root@sp asterisk]# free -m total used free sharedbuffers cached Mem: 3923 1934 1989 0270 1309 -/+ buffers/cache:353 3569 Swap: 4031 0 4031 So total of 3923 MB, where 1934 MB is used and 1989 MB still free. As I said, only a restart of Asterisk or reboot of the server can decrease this value : [root@sp asterisk]# free -m total used free sharedbuffers cached Mem: 3923974 2948 0150524 -/+ buffers/cache:299 3624 Swap: 4031 0 4031 So total of 3923 MB, where 974 MB is used and 2948 MB still free. So why does this memory usage only decrease with a restart ? What keeps asterisk stored in the RAM-memory ? Even if there are 0 calls, the memory does not decrease. Kind regards, 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 -- _ -- 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] Processed Call Counter
Thank you Eric for helping me out. I am using asterisk 1.6.x, 1.8.x
On Thu, Mar 8, 2012 at 9:48 PM, Eric Wieling ewiel...@nyigc.com wrote:
1.4:
pbx core show channels
[snip]
167 active channels
84 active calls
1.8:
pbx core show channels
[snip]
23 active channels
12 active calls
9567 calls processed
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, March 08, 2012 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Processed Call Counter
On Thu, 8 Mar 2012, [Digital^Dude] (r) wrote:
How can I reset the value of asterisk' calls processed without
restarting asterisk? Where does it save/access the value of all
processed calls since last restart from?
(I'm just a 1.2 Luddite, so my input may be a bit dated.)
Where are you seeing 'calls processed?'
There is a 'channels created' counter. That's the bit after the '.' in
${UNIQUEID}.
Resetting this is probably a bad idea, but if you're really determined you
could write a small Asterisk application (app_reset_call_counter()?) that
would lock the appropriate data structure, reset it, and unlock the data
structure.
Probably a better idea would be for you to maintain a counter in a global
variable. Then you are free to do with it as you please without having to
consider the implications of changing something internal to Asterisk.
If you stored your counter in a database, it could persist beyond that
instance of Asterisk in case of an unlikely crash. If you go this route, 2
AGIs (increment-call-counter and reset-call-counter) would help keep your
dialplan 'clean.'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Processed Call Counter
Danny, I use 1.6.x and 1.8.x asterisk versions. I would think asterisk call counters won't be changed in each version... hence I thought the asterisk version wouldn't be relevant. Reload of a particular application doesn't reset the counters. I have noticed that calls done with AMI Originate, don't get added in the asterisk counters (neither as active call, nor processed calls). On Thu, Mar 8, 2012 at 9:01 PM, Danny Nicholas da...@debsinc.com wrote: AFAIK, this is a “shell count” (The count is kept in shell memory for the running asterisk process). You handicap potential answer by not stating your Asterisk version or your technology (SIP/DAHDI/T1/etc). If you are just using SIP trunks, SIP RELOAD might do it. ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *[Digital^Dude] ® *Sent:* Thursday, March 08, 2012 2:41 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Processed Call Counter ** ** Hi there How can I reset the value of asterisk' calls processed without restarting asterisk? Where does it save/access the value of all processed calls since last restart from? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CDRs
I tried it on asterisk version 1.8 as well as other minor releases of 1.6. Its the same. Not a silly question, since silly products usually have instantaneous errors vanished by a silly action :) I tried changing different parameters in cdr.conf and reloading multiple times... Yes, I am just using the default CDR backend and I check CDR population via AST_LOG_DIR/Master.csv By the way, even if I use the mysql backend, its the same result. the CPU load is negligible and call quality is fine but still CDR write speed is almost 1 entry /1.5 second. I highly doubt that would be the max write speed! The kind of call burst I get on my asterisk box, tail -f Master.csv should be an unreadable scroll of entries! Hasn't *anyone *noticed it? On Fri, Mar 2, 2012 at 1:02 PM, Leandro Dardini ldard...@gmail.com wrote: Really interesting finding. From my point of view, it is a good thing. Having spike in cpu load will harm voice quality for sure, but it can hurts if you are relaying on prompt write of cdr records. What cdr backend are you using? Maybe the constant speed you see is the maximal write speed the backend can receive. A silly question ... have you reloaded the cdr module once made the changes? Leandro 2012/3/2 [Digital^Dude] ® millennium@gmail.com I've tried with batch enabled as well as disabled, it seems irrespective of the call burst I send to asterisk. CDR writes at a constant speed, not changing with the call load! On Fri, Mar 2, 2012 at 12:20 PM, Leandro Dardini ldard...@gmail.comwrote: Asterisk can cache cdr records to avoid having to write continuosly in the cdr backend. Writing in bunch instead one at once improves performance. Check the cdr.conf file and disable the option batch if it hurts you. Leandro Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com ha scritto: -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] AMI: Local Channels
Hello, I'm using Asterisk 1.6.2.10. Whenever I dial Local channels via asterisk manager, the calls never get a hangup signal even with timeout specified. I find channels with ZOMBIE text appended. It ends up occupying all the channels with the result that asterisk thinks every channel is busy, hence drops further calls. Also, all calls dialed out through the local channel, get the cdr populated before hangup (obviously with incorrect information). If someone else has gone through this problem please share and let me know how to rectify the issue. -- 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
[asterisk-users] SS7 Disposition
In almost all major releases of asterisk 1.6.x, SS7 Disposition never sets to ANSWERED, even when someone answers the call, it logs NO ANSWER in the cdrs. Please help me resolve the issue. -- 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] SS7 Disposition
What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SS7 Disposition
Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.comwrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SS7 Disposition
I tried it on asterisk 1.8, and it worked fine. On Thu, Mar 1, 2012 at 6:39 PM, Vinod Dharashive vdharash...@gmail.comwrote: ** Hi , Yes, I am using asterisk-java ami to originate call. Using LibSS7 Thanks Vinod dharashive Sent from BlackBerry® on Airtel -- *From: * [Digital^Dude] ® millennium@gmail.com *Date: *Thu, 1 Mar 2012 18:23:47 +0500 *To: *vdharash...@gmail.com; Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Subject: *Re: [asterisk-users] SS7 Disposition Are you using AMI originate for these SS7 outbound calls? On Thu, Mar 1, 2012 at 6:15 PM, [Digital^Dude] ® millennium@gmail.com wrote: What versions on Asterisk and chan_ss7 are you using? On Thu, Mar 1, 2012 at 3:50 PM, Vinod Dharashive vdharash...@gmail.comwrote: Hi team, I am experience the same issue. Thanks Vinod dharashive Sent from BlackBerry® on Airtel -Original Message- From: [Digital^Dude] ® millennium@gmail.com Sender: asterisk-users-boun...@lists.digium.com Date: Thu, 1 Mar 2012 15:32:41 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] SS7 Disposition -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CDRs
Hi all, It disturbs me to see asterisk (v 1.6.2.10) writing CDRs even when there are 0 active channels and 0 active calls. Is there an upper limit in terms of CDRs / second that asterisk can handle? Does it queue the unwritten CDRs somewhere? Please help me clarify this confusion. 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] Asterisk CDRs
I've tried with batch enabled as well as disabled, it seems irrespective of the call burst I send to asterisk. CDR writes at a constant speed, not changing with the call load! On Fri, Mar 2, 2012 at 12:20 PM, Leandro Dardini ldard...@gmail.com wrote: Asterisk can cache cdr records to avoid having to write continuosly in the cdr backend. Writing in bunch instead one at once improves performance. Check the cdr.conf file and disable the option batch if it hurts you. Leandro Il giorno 02/mar/2012 07:24, [Digital^Dude] ® millennium@gmail.com ha scritto: -- _ -- 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] Asterisk 1.8.x app_meetme.so
Hello, I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir. Also, I can find app_meetme change-logs on the asterisk website. However, the dialplan doesn't have this cmd. I have checked menuselect but it says it has been replaced by app_confbridge. Also, If that *is* the case, does ConfBridge (the newer version of meetme) offer the same options? How do I use them? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
I did, and I mentioned it in my earlier email too. Screenshot attached. On Wed, Feb 22, 2012 at 6:03 PM, Doug Lytle supp...@drdos.info wrote: [Digital^Dude] ® wrote: I can't seem to use MeetMe app in asterisk versions beyond 1.8. Its source file app_meetme.c is present in the apps dir It's still available, but marked as depreciated. So, do a make menuselect, Applications, scroll all the way to the bottom and check meetme. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users attachment: meetme.PNG-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
Doug, I can find the following in asterisk 10 changelogs: The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards This would mean that meetme should not have dahdi as a compilation dependency. source: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10-current On Wed, Feb 22, 2012 at 7:19 PM, Matthew Jordan mjor...@digium.com wrote: - Original Message - From: Doug Lytle supp...@drdos.info To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 22, 2012 7:22:20 AM Subject: Re: [asterisk-users] Asterisk 1.8.x app_meetme.so You mentioned that the meetme source was there, I was guessing that the option to compile wasn't checked so the binary wasn't available. I just ran into this myself yesterday when converting a 1.4x box (Still in progress) to a 10.2.0 RC2 and once checked and re-compiled, meetme was available. Doug Just a few points of clarification: 1. MeetMe is still the preferred conferencing application in Asterisk 1.8. In Asterisk 10, the preferred conferencing application is ConfBridge. Even still, in Asterisk, 10, you can compile and install MeetMe using menuselect. 2. In the screenshot you attached, you cannot choose to compile MeetMe as one of its dependencies is not available, in this case, DAHDI. Note that DAHDI being a dependency for MeetMe was one of the reasons Asterisk 10 moved to using ConfBridge as the default conferencing application. Matthew Jordan Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8.x app_meetme.so
So you mean I can't use dahdi_dummy with meetme? On Wed, Feb 22, 2012 at 9:28 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 02/22/2012 09:23 AM, [Digital^Dude] ® wrote: Doug, I can find the following in asterisk 10 changelogs: The following error will consistently occur when trying to dial into a MeetMe conference when the server does not have DAHDI hardware installed: app_meetme.c: No DAHDI channel available for conference, user introduction disabled (is chan_dahdi loaded?) While chan_dahdi is loaded correctly during compilation and install of Asterisk/Dahdi, including associated modules, etc., a chan_dahdi.conf configuration file in /etc/asterisk is not created by FreePBX if hardware does not exist, causing MeetMe to be unable to open a DAHDI pseudo channel. * Allow chan_dahdi to create a pseudo channel when there is no chan_dahdi.conf file to load. (closes issue ASTERISK-17398) Reported by: Preston Edwards This would mean that meetme should not have dahdi as a compilation dependency. No, this is incorrect. First, you are confusing DAHDI and chan_dahdi. MeetMe absolutely requires, and will always require DAHDI, because DAHDI is used for mixing the audio streams together into conferences. Second, MeetMe also requires chan_dahdi to be loaded, and prior to the patch you listed above, this required a chan_dahdi.conf file to be present. The patch listed above changed changed chan_dahdi to load in a very 'basic' configuration when no chan_dahdi.conf file is present. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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-**usershttp://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
