Re: [asterisk-users] Asterisk on DD-WRT : modules.conf not found
Thanks for your response. I have modified asterisk.conf as follow : [directories] astetcdir = /opt/etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /opt/var/lib/asterisk astdatadir = /opt/var/lib/asterisk astagidir = /opt/var/lib/asterisk/agi-bin astspooldir = /opt/var/spool/asterisk astrundir = /var/run astlogdir = /opt/var/log/asterisk But new problem arises when I start Asterisk (/opt/sbin/asterisk -c) : [Sep 30 08:57:06] == Manager registered action ParkedCalls [Sep 30 08:57:06] == Manager registered action Park [Sep 30 08:57:06] res_features.so = (Call Features Resource) [Sep 30 08:57:06] == Parsing '/opt/etc/asterisk/indications.conf': [Sep 30 08:57:06] Found Segmentation fault (core dumped) What causes a segmentation fault ?? Jonas. On Wed, 2009-09-30 at 00:07 +0200, Tzafrir Cohen wrote: Is /opt/etc/asterisk the compiled config directory (astetcdir)? What is the contents of asterisk.conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retrieve Call setup - QoS
Il giorno 29/set/09, alle ore 17:46, Danny Nicholas ha scritto:
I believe that this information is at least indirectly in the CDR.
[...]
If you subtract the 92 from the 97, you get the 5 second number
you’re looking for. These fields have actual names, but they aren’t
relevant to me since I’m using the flat-text CDR Master.csv.
I think that values are:
${CDR(start)} = time of the start of the call
${CDR(answer)} = time when the call was answered
but I want ${CDR(session progress or ringing)} instead of $
{CDR(answer)}: (time from SIP INVITE to 183 SESSION PROGRESS or 180
RINGING)
In addition, I don't know if ${CDR(start)} is the INVITE or RINGING
time...
Do you have any hint?
Thanks
___
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Re: [asterisk-users] Music On Hold
Hello, We posted the question below yesterday, but got no answer from the community. When we checked the same behavior with Asterisk 1.2, we got the Started music on hold, class... message on the console, but in 1.6, we get absolutely nothing. I tried to unload and reload the moh module and everything seems normal, but Asterisk still doesn't respond in the console to the HOLD action, represented by the INVITE message. the call itself is being placed on hold and can be retrieved, but the audio file is not played and the held party hears only a silence. If anyone knows how to debug/fix it, your help would be HIGHLY appreciated. We're really stuck. Thank you all in advance. Original Message Subject: Music On Hold From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, 29 September, 2009 14:31:28 Hello, We need help in debugging Music On Hold on our Asterisk 1.6.1.6 From the SIP debug, I see that an extension sends an INVITE of the call to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but I don't see in the console any reference to the call being placed on hold. When I typed moh show files, I see the wav files of the /var/lib/asterisk/moh folder. How can I debug this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Native bridging analog phones trouble DAHDI channels.
I've set transfer = no for all channels in chan_dahdi.conf, but I still have the same [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=no canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no ;faxdetect=incoming ;echotraining=800 callgroup=1 pickupgroup=1 relaxdtmf=yes This is the log of the second call. I am pressing flash to make the transfer, the bad thing is that a short on-hook situation simulate that flash, and are making this unwanted transfer. [Sep 30 07:17:41] VERBOSE[3237] logger.c: -- Called g2/16 [Sep 30 07:17:42] DEBUG[3237] chan_dahdi.c: Sent deferred digit string: T16w [Sep 30 07:17:43] VERBOSE[3237] logger.c: -- DAHDI/9-1 answered SIP/101-087c9288 [Sep 30 07:17:49] VERBOSE[3054] logger.c: -- Stopped music on hold on DAHDI/8-1 [Sep 30 07:17:49] DEBUG[3054] chan_sip.c: SIP transfer: Succeeded to masquerade channels. [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: New owner for channel 8 is DAHDI/8-1 [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: master: 8, slave: 9, nothingok: 0 [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Stopping tones on 8/0 talking to 9/0 [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Stopping tones on 9/0 talking to 8/0 [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Making 9 slave to master 8 at 0 [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Added 20 to conference 9/8 [Sep 30 07:17:49] DEBUG[3237] chan_dahdi.c: Added 19 to conference 9/9 Kevin P. Fleming escribi: Maurizio Faccio adinet wrote: I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. That's because you have just completed a flash-hook based transfer of the first call to the second call. If you don't want this feature, set 'transfer=no' for the relevant channels in chan_dahdi.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] put some IVR into a queue after the call queuing
Dear all is it possible to handle a queue using a programmed IVR? As i understood, is possible to: - do some IVR in the dialplan BEFORE to queue the call - put a timeout to exit from the call and then do some IVR in the dialplan - intercept a single dialtone to exit the queue and performe some IVR in the dialplan (context setting in the queue) I've tested these things but in each case if i re-queue the call thi queue_log file reports the wrong total queued time. I'm wondering if is possible to bluild a script like that: 1) queue the call 2) after x seconds prompt message A 3) after y seconds prompt message B 4) after z seconds prompt message C 5) after t seconds prompt message Z with DTMF options 1,2,3 if option is 1 = remain in queue if option is 2 = ask the user to be recalled if option is 3 = transfer to In each moment (1,2,3,4,5) if a member queue gets available the call is routed to him. I belive that the only thing to do that is to do something like: 1) Queue A ... timeount 2) Queue B ... timeout 3) Queue C ...Timeout 4) Queue D ...periodic-announce - context set to xxx [xxx] 1,1,Queue D 2,1,Goto (.IVR to be recalled) 3,1,Goto ( transfer) And then manually match information between unique ID and queue_log to consider info on queue A,B,C,D, as a single queue. Or is there some magic sauce to specify an IVR script that is executed when a call is in a queue? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
I'm afraid I can't be much help as I am both a newbie and it works just fine for me on 1.6.1.6. Of course, mine was a fresh installation. Is there anything in the logs to give you a clue? You see the wav files but do you see the files encoded for the codecs you are using? I think Asterisk will transcode on the fly but I'm not sure. Sorry - John On Wed, 2009-09-30 at 11:52 +0300, Cyprus VoIP wrote: Hello, We posted the question below yesterday, but got no answer from the community. When we checked the same behavior with Asterisk 1.2, we got the Started music on hold, class... message on the console, but in 1.6, we get absolutely nothing. I tried to unload and reload the moh module and everything seems normal, but Asterisk still doesn't respond in the console to the HOLD action, represented by the INVITE message. the call itself is being placed on hold and can be retrieved, but the audio file is not played and the held party hears only a silence. If anyone knows how to debug/fix it, your help would be HIGHLY appreciated. We're really stuck. Thank you all in advance. Original Message Subject: Music On Hold From: Cyprus VoIP voi...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, 29 September, 2009 14:31:28 Hello, We need help in debugging Music On Hold on our Asterisk 1.6.1.6 From the SIP debug, I see that an extension sends an INVITE of the call to the Asterisk, whenever the HOLD or Transfer buttons are pressed, but I don't see in the console any reference to the call being placed on hold. When I typed moh show files, I see the wav files of the /var/lib/asterisk/moh folder. How can I debug this? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
I'm afraid I can't be much help as I am both a newbie and it works just fine for me on 1.6.1.6. Of course, mine was a fresh installation. Thanks for your help, John. Mine is also a fresh installation, but now at least I know it's not a version issue. Is there anything in the logs to give you a clue? There's absolutely nothing in the logs, and that's what surprises me. You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music On Hold
On Wed, 2009-09-30 at 14:57 +0300, Cyprus VoIP wrote: snip You see the wav files but do you see the files encoded for the codecs you are using? There's only one wav file there. No encoded files, but on asterisk 1.2 we have, it's the same file and it works. snip Hmm . . only one wav file. We had several. As I recall now, we actually installed 1.6.1.1 and upgraded. 1.6.1.1 had the old hold music. 1.6.1.6 has the new hold music. But I believe there are several files. Is that wav file valid, i.e., if you copy it to a system with a sound card and play it, does it play? Could it have been corrupted in copying or have incorrect permissions? - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI channel congested busy
Shaun Ruffell wrote: On 09/29/2009 06:52 AM, Jerry Geis wrote: A user report that this issue: https://issues.asterisk.org/view.php?id=15429 Has resolved their problem with a TDM card. My card is a T1/PRI card. Different module to load. I have the same issue. Does this same problem exist in the PRI code and needs fixed their also? Has it been fixed? and does this issue warrant a new release? Unfortunately, if you're seeing this with a PRI code, it would be completely unrelated to issue 15429. Do you see anything interesting when you enable pri intense debug and try to make an outbound call? Shaun, When I login to the cli and type pri intense debug 1 for span 1 it says no such command pri intense debug 1 and type help pri intense for help. I do type help pri intense and it says a valid command pri intense debug span. What am I not getting? Thanks, Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to finish a Meetme
Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludueña _ Descubre todas las formas en que puedes estar en contacto con amigos y familiares. http://www.microsoft.com/windows/windowslive/default.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UpdateConfig
Thanks, It worked, it seems there was something wrong. The following is working now: Action: UpdateConfig srcFileName: voicemail.conf dstFileName: voicemail.conf Action-00: Append Cat-00: default Var-00: 2000 Value-00: ,Jhon ActionID: 1234 Bye, Anahi Ludueña Date: Tue, 29 Sep 2009 17:50:05 -0500 From: jsm...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] UpdateConfig - Danny Nicholas da...@debsinc.com wrote: Two questions: 1. do you need an ActionID line? Danny, It's *always* considered best practice to have an ActionID line in AMI commands, so that you can easily differentiate the responses, especially to asynchronous commands. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Chatea sin límites en Messenger con la tarifa plana de Orange http://serviciosmoviles.es.msn.com/messenger/orange.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Retrieve Call setup - QoS
I'm probably wrong, but IMO CDR{start} is the SIP Invite time and
CDR(answer) is the time that the 183 signal was received. You can probably
tweak sip.conf to make this so (or not).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlo Dimaggio
Sent: Wednesday, September 30, 2009 3:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Retrieve Call setup - QoS
Il giorno 29/set/09, alle ore 17:46, Danny Nicholas ha scritto:
I believe that this information is at least indirectly in the CDR.
[...]
If you subtract the 92 from the 97, you get the 5 second number
you're looking for. These fields have actual names, but they aren't
relevant to me since I'm using the flat-text CDR Master.csv.
I think that values are:
${CDR(start)} = time of the start of the call
${CDR(answer)} = time when the call was answered
but I want ${CDR(session progress or ringing)} instead of $
{CDR(answer)}: (time from SIP INVITE to 183 SESSION PROGRESS or 180
RINGING)
In addition, I don't know if ${CDR(start)} is the INVITE or RINGING
time...
Do you have any hint?
Thanks
___
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___
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Re: [asterisk-users] Asterisk on DD-WRT : modules.conf not found
On Wed, Sep 30, 2009 at 10:03:34AM +0200, jonas kellens wrote: Thanks for your response. I have modified asterisk.conf as follow : [directories] astetcdir = /opt/etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /opt/var/lib/asterisk astdatadir = /opt/var/lib/asterisk astagidir = /opt/var/lib/asterisk/agi-bin astspooldir = /opt/var/spool/asterisk astrundir = /var/run astlogdir = /opt/var/log/asterisk But new problem arises when I start Asterisk (/opt/sbin/asterisk -c) : [Sep 30 08:57:06] == Manager registered action ParkedCalls [Sep 30 08:57:06] == Manager registered action Park [Sep 30 08:57:06] res_features.so = (Call Features Resource) [Sep 30 08:57:06] == Parsing '/opt/etc/asterisk/indications.conf': [Sep 30 08:57:06] Found Segmentation fault (core dumped) What causes a segmentation fault ?? Not sure. But it's likely the module loaded after res_features.so . -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on pri intense debug
Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
pri intense debug span Enables REALLY INTENSE PRI debugging add span keyword or use a tabulator that will do that for you Martin On Wed, Sep 30, 2009 at 10:08 AM, Jerry Geis ge...@pagestation.com wrote: Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
Because you need to type pri intense debug SPAN 1 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Wednesday, September 30, 2009 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on pri intense debug Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
Jerry Geis wrote: Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? pri intense debug span span number -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
pri intense debug span span number Just pointing out that was not clear from the HELP command. I thought span was the span number not span span number Thanks for the direction. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PBXNSIP Registration Issue
I've got PBXNSIP running on a windows box and it is trying to register with my Asterisk box. I can set up one trunk and it works fine, however if I try to setup a second trunk from the same box, there is some sort of authentication issue where Asterisk appears to be confusing which trunk is which. Here is the chunk from my sip.conf: [TEST1] context=STUFF-LD type=friend callerid=TEST1 username=TEST1 secret= host=dynamic nat=yes canreinvite=no qualify=yes [TEST2] context=STUFF-LD type=friend callerid=TEST2 username=TEST2 secret= host=dynamic nat=yes canreinvite=no qualify=yes 'sip show peers' shows both registered on Asterisk ok. If I try and call out test2, it works. However, if I try and call out test1, it fails with this: [Sep 30 12:01:10] WARNING[16678]: chan_sip.c:8272 check_auth: username mismatch, have TEST2, digest has TEST1 [Sep 30 12:01:10] NOTICE[16678]: chan_sip.c:13587 handle_request_invite: Failed to authenticate user 3210 sip:3...@192.168.100.72;user=phone; tag=9055 What is happening is that since both regs come from the same remote IP, Asterisk thinks the call is coming from test2, even though it is really coming from test1 per the sip debug below. Any idea how to make Asterisk realize that the call is on test1? --- SIP read from 192.168.100.98:5060 --- INVITE sip:6...@192.168.100.72;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;rport From: 3210 sip:3...@192.168.100.72;user=phone;tag=63019 To: sip:6...@192.168.100.72;user=phone Call-ID: 26f91...@pbx CSeq: 23974 INVITE Max-Forwards: 70 Contact: sip:te...@192.168.100.98:5060;transport=udp Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 P-Asserted-Identity: TEST1 sip:te...@192.168.100.72 Content-Type: application/sdp Content-Length: 196 v=0 o=- 30939 30939 IN IP4 192.168.100.98 s=- c=IN IP4 192.168.100.98 t=0 0 m=audio 63088 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv - --- (16 headers 10 lines) --- Sending to 192.168.100.98 : 5060 (NAT) Using INVITE request as basis request - 26f91...@pbx Found peer 'TEST2' --- Reliably Transmitting (NAT) to 192.168.100.98:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;received =192.168.100.98;rport=5060 From: 3210 sip:3...@192.168.100.72;user=phone;tag=63019 To: sip:6...@192.168.100.72;user=phone;tag=as6350df4b Call-ID: 26f91...@pbx CSeq: 23974 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7edf0cb1 Content-Length: 0 Scheduling destruction of SIP dialog '26f91...@pbx' in 1344 ms (Method: INVITE) Retransmitting #1 (NAT) to 192.168.100.98:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;received =192.168.100.98;rport=5060 From: 3210 sip:3...@192.168.100.72;user=phone;tag=63019 To: sip:6...@192.168.100.72;user=phone;tag=as6350df4b Call-ID: 26f91...@pbx CSeq: 23974 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=7edf0cb1 Content-Length: 0 --- dell860*CLI --- SIP read from 192.168.100.98:5060 --- ACK sip:6...@192.168.100.72;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-b6ab2347b7aeeed06dbb606b08ea8b68;rport From: 3210 sip:3...@192.168.100.72;user=phone;tag=63019 To: sip:6...@192.168.100.72;user=phone;tag=as6350df4b Call-ID: 26f91...@pbx CSeq: 23974 ACK Max-Forwards: 70 Content-Length: 0 - --- (8 headers 0 lines) --- dell860*CLI --- SIP read from 192.168.100.98:5060 --- INVITE sip:6...@192.168.100.72;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.98:5060;branch=z9hG4bK-757d8077f9a2b9108ec158f2fc07d30a;rport From: 3210 sip:3...@192.168.100.72;user=phone;tag=63019 To: sip:6...@192.168.100.72;user=phone Call-ID: 26f91...@pbx CSeq: 23975 INVITE Max-Forwards: 70 Contact: sip:te...@192.168.100.98:5060;transport=udp Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/3.4.0.3201 P-Asserted-Identity: TEST1 sip:te...@192.168.100.72 Proxy-Authorization: Digest realm=asterisk,nonce=7edf0cb1,response=5ececd40c28f0378503e2dd6ee5cef14 ,username=TEST1,uri=sip:6...@192.168.100.72;user=phone,algorithm=MD5 Content-Type: application/sdp Content-Length: 196 v=0 o=- 30939 30939 IN IP4 192.168.100.98 s=- c=IN IP4 192.168.100.98 t=0 0 m=audio 63088 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv -
Re: [asterisk-users] question on pri intense debug
On Wednesday 30 September 2009 11:54:11 Jerry Geis wrote: pri intense debug span span number Just pointing out that was not clear from the HELP command. I thought span was the span number not span span number Thanks for the direction. At the list level, we only provide the keywords. If you had explicitly requested the individual syntax, you would have seen the complete command structure: *CLI help pri intense debug span Usage: pri intensive debug span span Enables debugging down to the Q.921 level *CLI -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] EXTENSION_STATE Asterisk 1.6
Hi I've a queue which has generic zap extensions (of my legacy PBX which is connected to asterisk via cross over on span 4 ) logged in ..The legacy pbx extensions are able to logon to queue perfect.. but Whenever a call comes in queue the status of that extension in queue show queuename always shows as NOT IN USE instead of ringing or In use as shown in a SIP extension..My question is there anyway to register custom generic zap extensions onto the hints and get the status via extension state command ..alternatviley how can I show the device status as In USE when that legacy extension is on a call ?? Requesting for a help Thanks Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] EXTENSION_STATE Asterisk 1.6
How do these extensions show up on a core show channels verbose? I do my hints like this [internal] - exten = 501,hint,SIP/100 - exten = 502,hint,DAHDI/1 - exten = 503,hint,ZAP/1 you should be able to register a hint based on the cscv output. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sriram Sent: Wednesday, September 30, 2009 1:52 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] EXTENSION_STATE Asterisk 1.6 Hi I've a queue which has generic zap extensions (of my legacy PBX which is connected to asterisk via cross over on span 4 ) logged in ..The legacy pbx extensions are able to logon to queue perfect.. but Whenever a call comes in queue the status of that extension in queue show queuename always shows as NOT IN USE instead of ringing or In use as shown in a SIP extension..My question is there anyway to register custom generic zap extensions onto the hints and get the status via extension state command ..alternatviley how can I show the device status as In USE when that legacy extension is on a call ?? Requesting for a help Thanks Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more room in scheduler
Are you using a VPM module? The dahdi changelog mentions some recent work related to VPM modules and HDLC aborts. https://issues.asterisk.org/view.php?id=15498 https://issues.asterisk.org/view.php?id=15529 I just rebuilt a server this weekend for the same problem on a single span card with a VPM. I usually have to restart asterisk to fix it, but I just noticed an instance in the logs where it recovered on its own a minute later: [2009-09-29 01:12:20] NOTICE[5290] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [2009-09-29 01:19:31] ERROR[5290] chan_dahdi.c: No more room in scheduler [2009-09-29 01:19:31] ERROR[5290] chan_dahdi.c: Asked to delete sched id -1??? [2009-09-29 01:19:31] ERROR[5290] chan_dahdi.c: No more room in scheduler --snip-- [2009-09-29 01:20:25] ERROR[5290] chan_dahdi.c: No more room in scheduler [2009-09-29 01:20:25] VERBOSE[5290] logger.c: == Primary D-Channel on span 1 down [2009-09-29 01:20:25] WARNING[5290] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-09-29 01:20:25] VERBOSE[5290] logger.c: == Primary D-Channel on span 1 up [2009-09-29 01:20:25] ERROR[5290] chan_dahdi.c: !! Got a UA, but i'm in state 7 [2009-09-29 01:20:26] VERBOSE[5290] logger.c: == Primary D-Channel on span 1 down [2009-09-29 01:20:26] WARNING[5290] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-09-29 01:20:26] ERROR[5290] chan_dahdi.c: !! Got S-frame while link down [2009-09-29 01:20:26] ERROR[5290] chan_dahdi.c: !! Got S-frame while link down [2009-09-29 01:20:26] ERROR[5290] chan_dahdi.c: !! Got S-frame while link down [2009-09-29 01:20:26] ERROR[5290] chan_dahdi.c: !! Got S-frame while link down [2009-09-29 01:20:26] VERBOSE[5290] logger.c: == Primary D-Channel on span 1 up I also spotted some similar log entries the day before, but surprisingly without a crash afterward: [2009-09-28 01:21:59] NOTICE[5290] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 [2009-09-28 01:22:01] ERROR[5290] chan_dahdi.c: ACK received for '0' outside of window of '20' to '21', restarting [2009-09-28 01:22:01] VERBOSE[5290] logger.c: == Primary D-Channel on span 1 down [2009-09-28 01:22:01] WARNING[5290] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-09-28 01:22:01] VERBOSE[5290] logger.c: == Primary D-Channel on span 1 up [2009-09-28 01:22:01] ERROR[5290] chan_dahdi.c: !! Got a UA, but i'm in state 7 I get the crash in asterisk 1.6.0.15 and 1.6.1.6 with dahdi 2.2.0.2, asterisk 1.4.26.2 with zaptel, on Centos 4.8 and Centos 5.3. It always happens around the same time (probably the telco running tests as you mentioned), and I always get 99% on dahdi_test. I'm scheduling a nightly restart for now, but I'm also considering ditching the VPM for a while. Marc Smith wrote on 09/18/2009 01:33:11 PM: Hi, I running into the following problem on my Asterisk setup: --snip-- [Sep 3 01:40:59] NOTICE[9170] chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 3 [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:07] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:08] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: No more room in scheduler [Sep 3 01:47:09] ERROR[9170] chan_dahdi.c: Asked to delete sched id -1??? --snip-- This happens once a week, at same about the same time (give or take a couple minutes). Always from span 3 too. It just continually spits out those messages until I restart Asterisk. I've seen others post about this, but haven't seen a real answer. Someone said to run a 'dahdi_test -v' when this happens; I did and I get 99% every time. Someone else said this is usually caused by the telco. running some type of test on the line, and I would agree since it happens every week at pretty much the same time and same day. So, yes, lets say the telco. is sending some type of
[asterisk-users] SIPAddHeader into the SDP?
I use SIPAddHeader today to put some proprietary info into the SIP header of an outbound call. Now I'd like to add some proprietary info to the SDP portion of an outbound call. Can this be done with SIPAddHeader? Thanks in advance, Tom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question on pri intense debug
Try 'pri intense debug span 1' Used it last night. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Thursday, 1 October 2009 4:09 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] question on pri intense debug Running asterisk 1.4.26.2 help pri pri debug span Enables PRI debugging on a span pri intense debug span Enables REALLY INTENSE PRI debugging pri no debug span Disables PRI debugging on a span pri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settings pri show spans Displays PRI Information pri show span Displays PRI Information pri show version Displays version of libpri pri unset debug file Ends PRI debug output to file then I type the following command: pri intense debug 1 No such command 'pri intense debug 1' (type 'help pri intense' for other possible commands) Why is it not understanding my command? Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish a Meetme
Anahi Ludueña wrote: Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, From the asterisk cli 'core show application MeetMe': User can exit the conference by hangup, or if the 'p' option is specified, by pressing '#'. Then the dialplan resumes, but why would you need to kick that user from the MeetMe? AFAIK there is no -easy- way to automatically kick out the last user from the conference when it is the only one left. What are you using MeetMe for? Saludos -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] kill sip user
Bayardo Sanchez wrote: I have a user but I need to give that user only kill and disable all connection cut calls what is the command in the CLI Please rephrase your question. I've just read your message 5 times and I still don't understand what do you want to do. Regards. PS: A 15+ line signature for a 2 line message is likely to upset many people on any mailing list. -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected to tapped lines. This means that you don't have to install anything in the PBX's/telco equipment that is going to be monitored, all you need is to install a device like the PN 633 Tap Connection Adapter that is available for example, from Sangoma, however I am sure there must be other vendors out there offering similar devices. Then you need to pull a pair of cables out of the adapter to your monitoring system with Sangoma boards configured in high impedance mode (I don't know if Digium or other vendors boards expose that functionality to users, but you may want to test and find out if works). More detailed instructions can be found at Sangoma's site or my blog: http://wiki.sangoma.com/sangoma-tap-system http://www.moythreads.com/wordpress/2009/09/26/sangoma-tapping-solution-for-asterisk/ The patches are already out there in the bug tracker along with some SVN branches. https://issues.asterisk.org/view.php?id=15970 https://issues.asterisk.org/view.php?id=15971 I'd love to get feedback in the bug tracker in order to get this feature into Asterisk soon :-) Also don't hesitate in asking for help with the configuration. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish a Meetme
there is an undocumented feature in meetme using the kick option called all, which kicks everyone off if you want to be sure and end the conference. Ivan Stepaniuk i...@albafotonica.com wrote: Anahi Ludueña wrote: Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, From the asterisk cli 'core show application MeetMe': User can exit the conference by hangup, or if the 'p' option is specified, by pressing '#'. Then the dialplan resumes, but why would you need to kick that user from the MeetMe? AFAIK there is no -easy- way to automatically kick out the last user from the conference when it is the only one left. What are you using MeetMe for? Saludos -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPAddHeader into the SDP?
Tom Browning wrote: I use SIPAddHeader today to put some proprietary info into the SIP header of an outbound call. Now I'd like to add some proprietary info to the SDP portion of an outbound call. Can this be done with SIPAddHeader? Nope; there is no way to make modifications to the SDP content of SIP messages in Asterisk without modifying chan_sip. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish a Meetme
On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote: there is an undocumented feature in meetme using the kick option called all, which kicks everyone off if you want to be sure and end the conference. Are you referring to the documented 'K' option for the meetmeadmin() dialplan application or the inadequately documented meetme kick confno usernumber CLI command -- which doesn't (1.2) document that usernumber can be all? (Or that confno does not have to be numeric.) -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish a Meetme
Steve Edwards asterisk@sedwards.com wrote: On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote: there is an undocumented feature in meetme using the kick option called all, which kicks everyone off if you want to be sure and end the conference. Are you referring to the documented 'K' option for the meetmeadmin() dialplan application or the inadequately documented meetme kick confno usernumber CLI command -- which doesn't (1.2) document that usernumber can be all? (Or that confno does not have to be numeric.) The cli command. I wish you could some of this from the phone, but you'd almost have to have an audio display of user numbers and caller ids to have it make sense. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici cov...@ccs.covici.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
Moises, You forgot to add that in order to monitor one T1 or E1 circuit you need two ports on your card... So that might be getting expensive with Sangoma cards You can do the same with cheap Tormenta cards that sell for ~$350 (I did that some time ago) Anyways all zaptel/dahdi cards can be set to high impedance since all the framers support it... However I'm pretty much sure it's not part of the drivers as of now. I had to enable the high impedance mode in the tormenta driver for myself for tests... Is your code vendor locked to Sangoma ??? Martin On Wed, Sep 30, 2009 at 3:12 PM, Moises Silva moises.si...@gmail.com wrote: Howdy, I've spent a couple of days writing a new feature for Asterisk that allows to record calls in T1 or E1 PRI lines using Asterisk connected to tapped lines. This means that you don't have to install anything in the PBX's/telco equipment that is going to be monitored, all you need is to install a device like the PN 633 Tap Connection Adapter that is available for example, from Sangoma, however I am sure there must be other vendors out there offering similar devices. Then you need to pull a pair of cables out of the adapter to your monitoring system with Sangoma boards configured in high impedance mode (I don't know if Digium or other vendors boards expose that functionality to users, but you may want to test and find out if works). More detailed instructions can be found at Sangoma's site or my blog: http://wiki.sangoma.com/sangoma-tap-system http://www.moythreads.com/wordpress/2009/09/26/sangoma-tapping-solution-for-asterisk/ The patches are already out there in the bug tracker along with some SVN branches. https://issues.asterisk.org/view.php?id=15970 https://issues.asterisk.org/view.php?id=15971 I'd love to get feedback in the bug tracker in order to get this feature into Asterisk soon :-) Also don't hesitate in asking for help with the configuration. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk over CentOS the module for Digium TE121 is not in the zaptel file
Hello I have a CentOS OS that have asterisk installed, also zaptel, but when I use the: lspci command I have the next asnwer: 03:80.0 Ethernet controller: Unknown device d161:8000 (rev 11) I also check the zaptel file that contain the modules that can support and the wcte12xp module is not in the file, so I think the problem is that the driver is not install into the OS. I know that we can migrate to dahdi, but at this time I need a zaptel file that can support this card, does anyone can help me with this issue? Thanks a lot for your help. Jhon Teleperformance values: Integrity - Respect - Professionalism - Innovation - Commitment The information contained in this communication is privileged and confidential. The content is intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify me immediately by telephone or e-mail, and delete this message from your systems. Please consider the environmental impact of needlessly printing this e-mail.___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to finish a Meetme
On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote: there is an undocumented feature in meetme using the kick option called all, which kicks everyone off if you want to be sure and end the conference. Steve Edwards asterisk@sedwards.com wrote: Are you referring to the documented 'K' option for the meetmeadmin() dialplan application or the inadequately documented meetme kick confno usernumber CLI command -- which doesn't (1.2) document that usernumber can be all? (Or that confno does not have to be numeric.) On Wed, 30 Sep 2009, cov...@ccs.covici.com wrote: The cli command. I wish you could some of this from the phone, but you'd almost have to have an audio display of user numbers and caller ids to have it make sense. I did this a couple of months ago. An admin, wanting to kick a user from a conference would execute an AGI that would map an index to a meetme user id via AMI so the admin could mute or un-mute a user (to identify the abusive user) or kick the user. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls at 2 different locations
I want to use IPKall with Asterisk. Now, I want my calls to land at 2 different locations , not connected with each other. If I want to configure IPKall DID number in Asterisk , I need to specify IP on IPKall. How can I make it enable so that calls can land up at both locations ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Choose IAX or SIP trunking?
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? I can name the provider if this is not against this list policy--is it? Thanks, -kkm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choose IAX or SIP trunking?
On Wed, 30 Sep 2009, Kirill 'Big K' Katsnelson wrote: Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID calls, originating and transferring. A provider offers both SIP and IAX trunking. Cateris paribus, what is the preferred solution to choose? What points to consider? Ceteris paribus, I prefer IAX. It tends to just work and it has a lot fewer knobs to turn. Some say the audio quality is better with SIP. My experience has been with low volume (xx) calls across the internet and high volume (xxx) within the same cabinet. I'd try IAX since it is so simple to configure. If you are not satisfied, try SIP. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
Is your code vendor locked to Sangoma ??? Hello Martin, not at all. The code is intended to be part of chan_dahdi Asterisk channel driver and as such any card capable of using the dahdi interface can benefit from it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chanspy and DISA
Steve Edwards wrote:
Steve Edwards wrote:
Is the manager or are the agents using disa()?
How about:
exten = *,n,set(SPYGROUP=ALLOW-SPYING)
for the agents and:
exten = *,n,chanspy(,g(ALLOW-SPYING))
the manager?
On Tue, 29 Sep 2009, John Millican wrote:
The manager wants to be able to spy on agents who dial through the PBX
from their homes. Currently the agents dial the main number, use the
secret code to get to authenticate and DISA, and then dial back out
for their sales calls. I have chanspy working great on all internal
phones/extensions use group to limit who can spy and who can not. It not
so much to allow spying it is finding the correct channel to spy on for
the remote users.
How about something like these snippets:
[i](!)
exten = i,1,goto(${CONTEXT},s,1)
[s](!)
exten = s,1,verbose(1,[${CONTEXT}:${EXTEN}])
[home-agent-login](i,s)
exten = s,n,read(AGENT-ID,enter-agent-number)
exten = s,n,set(SPYGROUP=${AGENT-ID})
.
.
.
[supervisor-login](i,s)
exten = s,n,read(AGENT-ID,enter-agent-number)
exten = s,n,chanspy(,g(${AGENT-ID}))
exten = s,n,goto(s,1)
.
.
.
Thank you very much for this.
With a little tweaking it worked great, since each remote workers
callerid is matched before going to authenticate I just set the spy
group so the remote guys don't have a choice and now the manager has a
known group of one for each remote worker.
Thanks again for the help
JohnM
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Re: [asterisk-users] Choose IAX or SIP trunking?
Steve Edwards wrote: Some say the audio quality is better with SIP. My experience has been with low volume (xx) calls across the internet and high volume (xxx) within the same cabinet. My understanding was that IAX encapsulates the same RTP traffic, or, and the very least, same stream of data encoded by a codec. Is that not true in case of IAX? How can a transport protocol affect volume--or quality (lest it is dropping packets)? -kkm, now puzzled. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E1/T1 Tapping call recording in Asterisk - Testing needed
That's nice. At least now peopel that want to do call recording can do so without having to keep Asterisk in between the circuits. However all other applications like added voicemail, conferencing, followme etc ... still needs Asterisk in between unless they have a spare port on the PBX and do the routing... Martin On Wed, Sep 30, 2009 at 7:47 PM, Moises Silva moises.si...@gmail.com wrote: Is your code vendor locked to Sangoma ??? Hello Martin, not at all. The code is intended to be part of chan_dahdi Asterisk channel driver and as such any card capable of using the dahdi interface can benefit from it. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. m...@sangoma.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choose IAX or SIP trunking?
Steve Edwards wrote: Some say the audio quality is better with SIP. My experience has been with low volume (xx) calls across the internet and high volume (xxx) within the same cabinet. On Wed, 30 Sep 2009, Kirill 'Big K' Katsnelson wrote: My understanding was that IAX encapsulates the same RTP traffic, or, and the very least, same stream of data encoded by a codec. Is that not true in case of IAX? How can a transport protocol affect volume--or quality (lest it is dropping packets)? My (limited) understanding is that IAX sends all call control and RTP to port 4569. Thus, a busy pipe can adversely affect timing if the single thread reading from the socket can't process the packets fast enough. Whether this manifests itself as dropped packets or jitter or whatever is beyond my experience. I've never had a client complain, but most of my traffic is within the same cabinet. -- 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with SIP QSIG phones in MeetMe conf room
My system is linked to a legacy PBX via Q-SIG and most of my tests so far have been from that PBX. I created a number of MeetMe conference rooms and they work fine when called from the legacy PBX. However, when there's a MeetMe room with a legacy caller and a SIP phone, the SIP phone can hear the legacy caller. But the legacy caller can't hear the SIP phone. However, meetme show conf does show the SIP caller as talking when they do. Here's the current channels when the conference is up in that configuration: asterisk*CLI core show channels Channel Location State Application(Data) DAHDI/23-1 2...@conferences:2Up MeetMe(201,cosT) DAHDI/pseudo-1338070 s...@default:1 Rsrvd (None) SIP/150-b444d988 2...@conferences:2Up MeetMe(201,cosT) What should I be looking at to debug this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] got stuck at 150 calls, above that not working in stress test
Hi All, I have a problem, when I was doing a performance testing using an asterisk server: Quadcore processor, 4GB RAM, CentOS5.2, after 150-151 calls all the other calls are giving busy, I tried to do ulimit related stuff, like increasing the soft and hard limits to 10 but no luck, Any ideas or views are really appreciated. Also I even changed the call limit to 500, but stills it can handle only 150 total. Thanks for your help. Regards Sandesh. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Choose IAX or SIP trunking?
Steve Edwards wrote: My understanding was that IAX encapsulates the same RTP traffic, or, and the very least, same stream of data encoded by a codec. Is that not true in case of IAX? How can a transport protocol affect volume--or quality (lest it is dropping packets)? My (limited) understanding is that IAX sends all call control and RTP to port 4569. Thus, a busy pipe can adversely affect timing if the single thread reading from the socket can't process the packets fast enough. Whether this manifests itself as dropped packets or jitter or whatever is beyond my experience. I've never had a client complain, but most of my traffic is within the same cabinet. I think I see now. Thanks for you response! -kkm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
