Re: [asterisk-users] test
On 25/10/2009, Matt mhop...@gmail.com wrote: This is a test... I am being told I am subscribed, but I am not getting messages. Gmail always seems to hide receipt of your own messages to mailing lists... Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help sip show on CLI : no such command
What is wrong when I can not execute any command that starts with sip ??? freepbx*CLI help sip show No such command 'sip show'. freepbx*CLI help sip No such command 'sip'. IAX works fine : freepbx*CLI help iax iax2 provision Provision an IAX device iax2 prune realtime Prune a cached realtime lookup iax2 reload Reload IAX configuration iax2 set debug Enable IAX debugging iax2 set debug jb Enable IAX jitterbuffer debugging iax2 set debug jb off Disable IAX jitterbuffer debugging iax2 set debug off Disable IAX debugging iax2 set debug trunk Enable IAX trunk debugging iax2 set debug trunk off Disable IAX trunk debugging iax2 show cache Display IAX cached dialplan iax2 show channels List active IAX channels iax2 show firmware List available IAX firmwares iax2 show netstats List active IAX channel netstats iax2 show peers List defined IAX peers iax2 show peer Show details on specific IAX peer iax2 show registry Display IAX registration status iax2 show stats Display IAX statistics iax2 show threads Display IAX helper thread info iax2 show users List defined IAX users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help sip show on CLI : no such command
On Sun, Oct 25, 2009 at 8:51 AM, jonas kellens jonas.kell...@telenet.be wrote: What is wrong when I can not execute any command that starts with sip ??? This often happens to me if for whatever reason my system doesn't have SIP support. Most of the time, this is caused by an error in sip.conf. Do a reload, and see if you get any errors around SIP? -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help sip show on CLI : no such command
On Sun, Oct 25, 2009 at 02:51:16PM +0100, jonas kellens wrote: What is wrong when I can not execute any command that starts with sip ??? freepbx*CLI help sip show No such command 'sip show'. freepbx*CLI help sip No such command 'sip'. IAX works fine : freepbx*CLI help iax iax2 provision Provision an IAX device iax2 prune realtime Prune a cached realtime lookup [snip] chan_sip.so is not loaded? What is the output of the following two CLI commands? module show like sip module load chan_sip.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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP interconnection problem
Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a Failed to authenticate on INVITE on the * to which the Zoiper is registered: -- Accepting AUTHENTICATED call from 192.168.10.113: Zoiper IP requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [010...@users:1] Dial(IAX2/2200-12940, SIP/010...@192.168.10.11) in new stack == Using SIP RTP CoS mark 5 -- Called 010...@192.168.10.11 Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '2200 sip:2...@192.168.10.77;tag=as3e4fedb8' 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2...@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk-stat! - help needed (once again due to mailserver problem)
Dear all, I'm trying to install Asterisk-stat (ASTERISK CDR ANALYSER) following: http://www.voip-info.org/wiki/index.php?page=Asterisk+CDR+Areski+GUI however it fails to run properly - lots of lines like: **Notice**: Undefined variable: s in */home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *26* **Notice**: Undefined variable: t in */home/lukasz/DATA/www/asterisk-stat/cdr.php* on line *27* See: http://i34.tinypic.com/16jja6s.jpg http://i34.tinypic.com/30u63qx.jpg My system: Asterisk 1.6.2.0-beta3, Apache 2.2.9, PHP: 5.2.6, MySQL Ver 14.12 Distrib 5.0.51a. Thanks for the advice! Cheers, Pepesz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FW: Queue Transfers
Hi Is there anyway to log all the transfers on an active call. I got a reply from my previous post that only the first transfer is logged and after that it is out of the queue application;s purview.. but my scenario is like agent A can transfer a caller to agent B and agent B to agent C .. on current asterisk queue implementation only the first transfer is logged ..i was also adviced to route the caller back to another queue which is an exact replica of the first queue so that irrespective of how many times a call is transferred all transfer events are logged in queue_log .. but my problem is that say Agent A picked the call first but the caller wants to talk to Agent C ..the Agent A cant call the duplicate queue as the call might land to another waiting Agent .. Is there any way to force a caller to land his call to a particular agent but through the Queue command ? Thanks Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Transfers
Hi Is there anyway to log all the transfers on an active call. I got a reply from my previous post that only the first transfer is logged and after that it is out of the queue application;s purview.. but my scenario is like agent A can transfer a caller to agent B and agent B to agent C .. on current asterisk queue implementation only the first transfer is logged ..i was also adviced to route the caller back to another queue which is an exact replica of the first queue so that irrespective of how many times a call is transferred all transfer events are logged in queue_log .. but my problem is that say Agent A picked the call first but the caller wants to talk to Agent C ..the Agent A cant call the duplicate queue as the call might land to another waiting Agent .. Is there any way to force a caller to land his call to a particular agent but through the Queue command ? Thanks Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP interconnection problem
you need to post you SIP.conf and your Extensions.conf so someone can have a look at them and see if there is anything missing what are the contexts you are using with your peers? what is the dial plan triggered when calling your destination number? -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sun, 25 Oct 2009 15:19:28 +0100 From: robert.bie...@xponaut.se To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP interconnection problem Hi all, I've setup two * servers which are SIP interconnected ala osaka/toronto from the * book (before anyone sugggests using IAX instead, no, I NEED to have them SIP interconnected for verification/test purposes). Then I have a Zoiper connected to one of them via IAX (so that * will not reinvite (?)). As soon as I try to call (via Zoiper) an extension on the other * I get a Failed to authenticate on INVITE on the * to which the Zoiper is registered: -- Accepting AUTHENTICATED call from 192.168.10.113: Zoiper IP requested format = gsm, requested prefs = (), actual format = ulaw, host prefs = (ulaw|alaw|gsm), priority = mine -- Executing [010...@users:1] Dial(IAX2/2200-12940, SIP/010...@192.168.10.11) in new stack == Using SIP RTP CoS mark 5 -- Called 010...@192.168.10.11 Other * [Oct 23 11:08:25] NOTICE[13576]: chan_sip.c:15031 handle_response_invite: Failed to authenticate on INVITE to '2200 sip:2...@192.168.10.77;tag=as3e4fedb8' 192.168.10.77 == * for Zoiper -- SIP/192.168.10.11-0a1716f8 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'IAX2/2200-12940' status is 'CONGESTION' -- Hungup 'IAX2/2200-12940' Why does * try to authenticate on sip:2...@192.168.10.77, it is IAX for crying out loud :) ? I've set canreinvite=no on the IAX phone (not sure this has any meaning in IAX at all) Not sure that this is root of the interconnection problem, since I then get SIP/192.168.10.11.. is circuit-busy... ? TIA /R ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows 7: I wanted more reliable, now it's more reliable. Wow! http://microsoft.com/windows/windows-7/default-ga.aspx?h=myidea?ocid=PID24727::T:WLMTAGL:ON:WL:en-US:WWL_WIN_myidea:102009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] help sip show on CLI : no such command
freepbx*CLI module show like sip Module Description Use Count chan_sip.soSession Initiation Protocol (SIP) 0 app_adsiprog.soAsterisk ADSI Programming Application 0 2 modules loaded freepbx*CLI module load chan_sip.so == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip_general_additional.conf': Found == Parsing '/etc/asterisk/sip_general_custom.conf': Found == Parsing '/etc/asterisk/sip_nat.conf': Found == Parsing '/etc/asterisk/sip_registrations_custom.conf': Found == Parsing '/etc/asterisk/sip_registrations.conf': Found == Parsing '/etc/asterisk/sip_custom.conf': Found == Parsing '/etc/asterisk/sip_additional.conf': Found == Parsing '/etc/asterisk/sip_custom_post.conf': Found == Parsing '/etc/asterisk/users.conf': Found == SIP Listening on 0.0.0.0:5060 == Using SIP TOS: CS3 == Parsing '/etc/asterisk/sip_notify.conf': Found == Parsing '/etc/asterisk/sip_notify_custom.conf': Found == Parsing '/etc/asterisk/sip_notify_additional.conf': Found == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered custom function SIP_HEADER == Registered custom function SIPPEER == Registered custom function SIPCHANINFO == Registered custom function CHECKSIPDOMAIN == Manager registered action SIPpeers == Manager registered action SIPshowpeer Loaded chan_sip.so = (Session Initiation Protocol (SIP)) Chan SIP was loaded, right ?? Loading the chan_sip again makes everything work now : freepbx*CLI sip show subscriptions Peer UserCall ID ExtensionLast state Type Mailbox 0 active SIP subscriptions On Sun, 2009-10-25 at 16:06 +0200, Tzafrir Cohen wrote: chan_sip.so is not loaded? What is the output of the following two CLI commands? module show like sip module load chan_sip.so ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OSLEC with DAHDI and Linksys/Sipura
On 10/25/09 02:35, Tzafrir Cohen wrote: On Fri, Oct 23, 2009 at 01:57:51PM -0500, Martin wrote: On Fri, Oct 23, 2009 at 5:41 AM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: Well, I've never used zaptel for Linsys/Sipura adapter as zaptel does not make any difference on their operations. and I'm not sure that DAHDI drivers exist for Linksys/Sipura adapters. DAHDI stands for Digium Asterisk Hardware Device Interface so I don't see how can I take advantage of DAHDI with Sipura adapters. Which is only because nobody wrote it yet. writing a DAHDI driver for a SIP device would be counter productive ... Re-flashing the device with your own custom Linux (OpenWRT?) and then providing DAHDI drivers for it isn't. The device in question is Linksys 3000 I don't think it is possible with this unit; if it was possible I'm sure someone would have done it; as the echo problem on PSTN line in these devices is terrible. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as the recording server for Avaya Definity
Has anyone tried to replace Witness or Nice recorder with asterisk. I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk. The idea behind is to record not only the external channels but also extension to extension (three way calling for which the third leg is asterisk PRI will do) Any suggestion will highly help Sam ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_echolink
Greetings, Where can I get the chan_echolink channel driver from? I've seen reference to it, but have yet to find a place to download/compile it. It is part of the app_rpt.so module... I am told, but do not see the source with app_rpt. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk as the recording server for Avaya Definity
Research wrote: . I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk. Could you send me the link of the article? I'll be looking into doing this within the next year. Thanks, 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] some issue with libpri cant go past 1.4.1
I have a working system with asterisk 1.4.26.2 libpri 1.4.1 and zaptel 1.4.12.1 With a digium TE205p. I am trying to update to libpri 1.4.10.2. When I do, incoming calls work but outgoing does not. When I do this I rm /usr/lib/libpri* then just install libpri-1.4.10.2 as normal. I then do a make clean in asterisk and make distclean ,then configure, make and make install. I do that exact same thing to get back to 1.4.1 as 1.4.10.2 does not allow me to call out. Below are the log files for both. I enabled pri intense debug span 1 for both a WORKING call and an ERROR call. I basically call into my system, selection option 2 which then calls out to my cell phone. What is the issue with 1.4.10.2? Thanks, (note I tried the same thing with other libpri versions). Jerry This is the error log: Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 044 P/F: 0 0 bytes of data -- Restarting T203 timer q931.c:2816 q931_call_proceeding: call 4 on channel 23 enters state 9 (Incoming Call Proceeding) -- Finally transmitting 40, since window opened up (0) [ 00 01 50 58 08 02 80 04 02 18 03 a9 83 97 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 040 0: 0 N(R): 044 P: 0 10 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 4/0x4) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] -- Accepting call from '3178672038' to '3175551212' on channel 0/23, span 1 -- Executing [3175551...@smvoice-incoming:1] Dial(Zap/23-1, SIP/a_to_b/3175551212) in new stack [Kmndemo*CLI -- Called a_to_b/3175551212 [Kmndemo*CLI [ 00 01 01 52 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 041 P/F: 0 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 39 to (but not including) 41 -- ACKing packet 40, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 timer [Kmndemo*CLI -- SIP/a_to_b-0174a060 answered Zap/23-1 q931.c:2951 q931_connect: call 4 on channel 23 enters state 8 (Connect Request) -- Finally transmitting 41, since window opened up (0) [ 00 01 52 58 08 02 80 04 07 18 03 a9 83 97 1e 02 81 82 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 041 0: 0 N(R): 044 P: 0 14 bytes of data Stopping T_203 timer Starting T_200 timer -- Restarting T200 timer Protocol Discriminator: Q.931 (8) len=14 [Kmndemo*CLI Call Ref: len= 2 (reference 4/0x4) (Terminator) Message type: CONNECT (7) [18 03 a9 83 97] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 23 ] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] [Kmndemo*CLI [ 00 01 01 54 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 042 P/F: 0 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 40 to (but not including) 42 -- ACKing packet 41, new txqueue is -1 (-1 means empty) -- Since there was nothing left, stopping T200 counter -- Nothing left, starting T203 counter -- Restarting T203 timer [Kmndemo*CLI [ 02 01 58 54 08 02 00 04 0f ] Informational frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 N(S): 044 0: 0 N(R): 042 P: 0 5 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 41 to (but not including) 42 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 4/0x4) (Originator) Message type: CONNECT ACKNOWLEDGE (15) q931.c:3711 q931_receive: call 4 on channel 23 enters state 10 (Active) Sending Receiver Ready (45) [ 02 01 01 5a ] Supervisory frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 045 P/F: 0 0 bytes of data -- Restarting T203 timer [Kmndemo*CLI [Oct 25 19:01:46] DTMF[20171]: channel.c:2262 __ast_read: DTMF end '2' received on Zap/23-1, duration 0 ms
Re: [asterisk-users] OT - mISDN and B410P questions
2009/10/24 Jean-Denis Girard jd.gir...@sysnux.pf -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier a écrit : | Hello, | | I'm evaluating to possibility to use chan_misdn as a short term | workaround, in case latest Dahdi is not stable enough for what we are | planning to do (we wish to use Junghanns and Digium BRI hardware with | Asterisk 1.6) . Dahdi has been working fine for me for a few months, using a Junghanns DuoBRI and asterisk-1.6. I have used bristuff before, it usually worked fine, but separate patch. I also tried chan_misdn but quickly abandoned. Do you mean you abondoned chan_misdn (1.1.9-1) with 1.6.x on Junghanns hardware or are you thinking about another combination ? Did you change because revision-2822-enabled Dahdi ran successfully enough that you didn't need to further try with misdn or because you met bloking issues ? Regards, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAkrjbg0ACgkQuu7Rv+oOo/hvxACggSiVlumqoSBtbcvUHKDHj1SG B8kAnicGGenJgcQXYq5RrkX7DCzYhpbR =jXB8 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
