Re: [asterisk-users] Help-ASTERISK-MFCR2
echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no musiconhold=default protocolclass=mfcr2 protocolvariant=ar,10,10 protocolend=cpe group = 1 context= e1-incoming channel = 1-15 channel = 17-31 ;skip time slot 16 Here is the LOGS when I try do make calls Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 0001 [1/ 1/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Detected Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Making a new call with CRN 32769 Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1101 - [2/ 2/Idle /Idle ] Jun 6 16:02:18 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Detected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 - 1001 [2/ 2/Seize ack /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Far end disconnected(cause=Normal, unspecified cause [31]) - state 0x2 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Far end disconnected Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2930 handle_uc_event: CRN 32769 - far disconnected cause=Normal, unspecified cause [31] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(6) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Drop call(cause=Normal Clearing [16]) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call disconnected(cause=Normal, unspecified cause [31]) - state 0x800 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Drop call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Call control(7) Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release call Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 1001 - [1/1000/Clear fwd /Seize ack] Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Release guard expired Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Destroying call with CRN 32769 Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:2644 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released Jun 6 16:02:30 WARNING[5060]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/1 Channel echo cancel Best Regards, Mariano Borgognone ___ -- 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 -- I do not agree with what you have to say, but I'll defend to the death your right to say it. Voltaire ___ -- 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 -- Alvaro I. Parres Peredo Director de IT Grupo Xmarts SA de CV Tel: +52 (33) 35 63 6261 Ext. 112 01 800 087 2260 Cel: +52 (33) 33 68 1087 [EMAIL PROTECTED] ___ -- 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 -- caio ___ -- 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 DTMF Tones
Same problem. I've tested a Linksys/PAP2-3.1.9(LSc) and tried with INBAND configuration in both, asterisk and linksys EP, and it works. But, just was a test, dont know if I would let it in INBAND config. Lastly I tried with INFO in linksys, and rfc2833 in Asterisk, and works too.., no problem. On 8/3/07, Keshav K. [EMAIL PROTECTED] wrote: I have used Asterisk 1.2 and 1.4 with ATAs and PAP2. There is no issue in that. For that confrim to your service provider that whihc they accepts, invand or rfc Keshav John Meksavan [EMAIL PROTECTED] wrote: Asterisk Users, I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having problems with DTMF Tones. I have sip service from Teliax and configure to use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T phone adapter with a regular analog phone. Is this an issue with Asterisk? Or the Linksys PAP2Any insights would be greatly appreciated. Best Regards, John _ http://liveearth.msn.com ___ --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 Regards, Kesh Lets change the future...lets change the world. Take the Internet to Go: Yahoo!Go puts the Internet in your pocket: mail, news, photos more. ___ --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 -- caio ___ --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] realtime ldap peer matching
On 04/06/07, Caio Zanolla [EMAIL PROTECTED] wrote: Hi everyone, in ldap realtime sip peers i need fullcontact set to sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least for the natted peers to reach them)... anyway, how do I populate fullcontact on the fly with information from exten and userip? Wouldn't these just be dialplan vars? What happens is when a natted client calls a non natted extension the debug shows chan_sip building the peer and res_ldap complaining about not finding fullcontact, then the calee is not found and the call is disconnected. If I manually add an attribute to the user w the correct fullcontact it goes ahead and finds it and completes the call. Note that fullcontact is not necessary for static peer, just realtime, cause if i try to reach a non nat exten (static) from a natted one it works flawlessly It has happened before as res_ldap was complaining about ipaddr, no problem since I already got it as a user attribute. Im puzzeld about peer building being so different for realtime. Not that I'm that familiar w normal peer building, but relatime is very picky. by dialplan vars you mean fullcontact on peer matching? well, i'd really like to know how to manipulate it. of course, i could just do it staticaly on ldap but since the info is already there why not make use of it? on res_ldap.conf i have attribute = fullcontact = AstAccountFullContact it would be nice to have something like: attribute = fullcontact = sip:.$AstExten.@.$AstIPaddress or some kind of dialplan scripting to archieve this... I'm pretty sure res_ldap.c can't do this yet. What version (* and res_ldap) and schema are you using btw? Im using * 1.4 svn checked out from res_ldap branch 4 weeks ago... IIRC, the latest version doesn't need: attribute = fullcontact = AstAccountFullContact just: fullcontact = AstAccountFullContact sure, it works both ways... Thanks, Gavin. cheers, Caio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime ldap peer matching
Hi everyone, in ldap realtime sip peers i need fullcontact set to sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least for the natted peers to reach them)... anyway, how do I populate fullcontact on the fly with information from exten and userip? of course, i could just do it staticaly on ldap but since the info is already there why not make use of it? on res_ldap.conf i have attribute = fullcontact = AstAccountFullContact it would be nice to have something like: attribute = fullcontact = sip:.$AstExten.@.$AstIPaddress or some kind of dialplan scripting to archieve this... cheers, Caio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Welltech FXO: initial tests
Dear Claudio, I'm testing the welltech gateways (3804 firmware 4fxosip.102) and I am trying to make the Asterisk answer the calls from 3508 directly (with 2nddial off) it means throw hotline service. Do you know how to make the Asterisk answer a call from :pstn-to-3508 and 3508-hotline-Asterisk ? Please let me know if the question is not clear enough ! My configurations are: extension.conf: [general]static=yeswriteprotect=no [default]include = oi exten = s,2,Answerexten = s,3,Wait(1)exten = s,4,Background(vm-toenternumber) ; Qual a extenssão desejada?exten = 1,1,Goto(oi)exten = 2,1,Goto(oi)exten = 3,1,Goto(oi)exten = 6,1,Goto(oi)exten = 11,1,playback(beep)exten = 0,1,Goto(default,s,2)exten = t,1,Goto(timeout,s,1)exten = i,1,Goto,s|2 [oi]exten = s,1,Wait(1)exten = s,2,Answerexten = s,3,Wait(1)exten = s,4,playback(pbx-transfer)exten = s,5,Goto(default,s,2) [bogon-calls]exten = _.,1,Congestion sip.conf [general]port=5060bindaddr=0.0.0.0context=from-sip;context=bogon-calls;context=defaultmaxexpirey=3600defaultexpirey=120disallow=allallow=ulawallow=alaw [300]port=5060type=friendcontext=defaultusername=9; Username to use in INVITE until peer registerssecret=fisa9host=10.150.3.100 disallow=allallow=ulawallow=alaw;allow=g729 3804: usr/config$ sip -print Run Mode : PROXY MODE Proxy server address : 10.150.3.4 Domain : null Prefix string : 1234 Line1 : 100 Line2 : 101 Line3 : 102 Line4 : 103 SIP port : 5060 RTP port : 16384 Expire : 3600 usr/config$ sysconf -print System information Inter-Digit time out : 1 End of Dial : No end of dial Port status: port1: Enabled port2: Enabled port3: Enabled port4: Enabled DTMF selection : In-band RFC2833 Payload Type : 96 FAX Payload Type : 101 2nddial: 3 Billing: OFF Dial Rule ip side: filter: [] drop: [] insert: []. pstn side: filter: [] drop: [] insert: []. PIN prompt: 0 set1: set2: set3: set4: Ring Detect Method: 1 Ring before Answer: 0 usr/config$ bureau -print Bureau line setting relate information PSTN number : 4198 2000 2001 2002 Hold tone generation : On Hot line / Line to Line table=Port Destination Address Remote TEL/CHANNEL-1 10.150.3.4 3002 300 3003 300 3004 300 300== ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
