[asterisk-users] how to write svn for dahdi-linux and dahdi-tools when using svn 1.4
Hi, When we use svn branches-1.4 such as: # svn checkout http://svn.digium.com/svn/asterisk/branches/1.4 asterisk-1.4 # svn checkout http://svn.digium.com/svn/libpri/branches/1.4 libpri-1.4 how to write the others such as dahdi-linux and dahdi-tools? Regards, Zen ___ -- 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] voicemail to non-default context user does not work
According to voip-info, the syntax for the VoiceMail command is as follows... VoiceMail([/flags/]/[EMAIL PROTECTED][EMAIL PROTECTED]boxnumber3]/) If you check the syntax for the VoiceMail command, it indicates that the mailbox parameter is /not/ optional, so I'm surprised this works at The context [03] worked well on asterisk-1.0.9. But recent version of asterisk i.e.,1.4.18 does not work. If I moved voicemail boxes to under default context, it works fine. I don't know why. all. Asterisk will default to the @default context if the context isn't specified, so you /might/ try Voicemail(@03) otherwise I suspect you're going to need an IVR to achieve what you want. exten = 0021,3,Voicemail(@03) does not work. -- Zen ___ -- 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] SOLVED::Re: voicemail to non-default context user does not work
Hi, I solved the problem. After I uncommented ; searchcontexts=yes in voicemail.conf, every things work fine. -- Zen ___ -- 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] voicemail to non-default context user does not work
Hi, I input 0203# after mailbox? voice prompt from Voicemail cmd on extensions.conf such as exten = 0021,1,Ringing exten = 0021,2,Wait(1) exten = 0021,3,Voicemail exten = 0021,4,Hangup *CLI -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/0103-09a308b0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/0103-09a308b0, 1) in new stack -- Executing [EMAIL PROTECTED]:3] VoiceMail(SIP/0103-09a308b0, ) in new stack -- SIP/0103-09a308b0 Playing 'vm-whichbox' (language 'en') [Feb 9 17:11:54] WARNING[3574]: app_voicemail.c:2850 leave_voicemail: No entry in voicemail config file for '0203' -- Executing [EMAIL PROTECTED]:4] Hangup(SIP/0103-09a308b0, ) in new stack == Spawn extension (sip, 0021, 4) exited non-zero on 'SIP/0103-09a308b0' -- Executing [EMAIL PROTECTED]:1] Hangup(SIP/0103-09a308b0, ) in new stack == Spawn extension (sip, h, 1) exited non-zero on 'SIP/0103-09a308b0' I have an entry of 0203 at Context 03 on voicemail.conf as follows; *CLI voicemail show users ContextMbox User Zone NewMsg defaultgeneral New User 0 03 01030 03 02030 03 03030 But, I could not enter into [EMAIL PROTECTED] mailbox. Any idea? asterisk-1.4.18 -- Zen ___ -- 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] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
Hi, I could found out why the phone received '404 Not Found'. The reason was this part is not parsed and not Added extensions after that. Because there was not at least one space after ; in front of the line of exten = 0033,1,Meetme(|qM). Regards, Zen From: Zen Kato [EMAIL PROTECTED] Subject: [asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found Date: Tue, 01 Aug 2006 12:15:04 +0900 (JST) Hi, I installed asterisk-1.2.10, zaptel-1.2.7 on 2.6.16-1.2108_FC4smp. When I dial '0033', which is a meetme number, but '404 Not Found' comes back. I checked zaptel(ztdummy) on FC4, it seems work fine. Meetme has been working on FC3. Can someone tell me why this happens on FC4? My extensions.conf is; exten = 0033,1,Meetme(|qM) exten = 0033,2,Hangup ngrep shows as follows; U 192.168.0.103:5060 - 192.168.0.3:5070 INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bKa854c86267e80f96..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3 fa865dcc1..To: sip:[EMAIL PROTECTED]:5070..Contact: sip:[EMAIL PROTECTED] 3..Supported: replaces..Call-ID: [EMAIL PROTECTED]: 589 86 INVITE..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: ap plication/sdp..Content-Length: 354v=0..o=0303 8000 8000 IN IP4 192.168. 0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m=audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=r tpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap:2 G726-32/8000..a=rtpmap :15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=20..a=rtpmap:9 G722/16 000..a=ptime:20.. # U 192.168.0.3:5070 - 192.168.0.103:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.103;b ranch=z9hG4bKa854c86267e80f96;received=192.168.0.103..From: sip:[EMAIL PROTECTED] 68.0.3:5070;tag=c7a5ee3fa865dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01 593a47..Call-ID: [EMAIL PROTECTED]: 58986 INVITE..User-A gent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCR IBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]:5070..Proxy-Authenticate: Dige st algorithm=MD5, realm=asterisk, nonce=72494d6d..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bKa854c86267e80f96..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3fa8 65dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Contact: sip:0303@ 192.168.0.103..Call-ID: [EMAIL PROTECTED]: 58986 ACK..U ser-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,C ANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bK6e9ddb4b834276ef..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3 fa865dcc1..To: sip:[EMAIL PROTECTED]:5070..Contact: sip:[EMAIL PROTECTED] 3..Supported: replaces..Proxy-Authorization: Digest username=0303, realm =asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=72494d6 d, response=35378e1d15e71946d8ca187b102d0087..Call-ID: 1c59a92f2174f5ca@ 192.168.0.103..CSeq: 58987 INVITE..User-Agent: Grandstream BT100 1.0.6.8..M ax-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUB SCRIBE..Content-Type: application/sdp..Content-Length: 354v=0..o=0303 8 000 8001 IN IP4 192.168.0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m =audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap: 2 G726-32/8000..a=rtpmap:15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mod e=20..a=rtpmap:9 G722/16000..a=ptime:20.. # U 192.168.0.3:5070 - 192.168.0.103:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.0.103;branch=z9hG4bK6e9ddb4 b834276ef;received=192.168.0.103..From: sip:[EMAIL PROTECTED]:5070;tag=c7a 5ee3fa865dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Call-ID: 1c5 [EMAIL PROTECTED]: 58987 INVITE..User-Agent: Asterisk PBX.. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact : sip:[EMAIL PROTECTED]:5070..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bK6e9ddb4b834276ef..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3fa8 65dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Contact: sip:0303@ 192.168.0.103..Proxy-Authorization: Digest username=0303, realm=asteris k, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=72494d6d, respo nse=9bea041787bf296bcd1c5d730733f615..Call-ID: [EMAIL PROTECTED] .103..CSeq: 58987 ACK..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS
[asterisk-users] FC4(2.6.16-1.2108_FC4smp) meetme 404 Not Found
Hi, I installed asterisk-1.2.10, zaptel-1.2.7 on 2.6.16-1.2108_FC4smp. When I dial '0033', which is a meetme number, but '404 Not Found' comes back. I checked zaptel(ztdummy) on FC4, it seems work fine. Meetme has been working on FC3. Can someone tell me why this happens on FC4? My extensions.conf is; exten = 0033,1,Meetme(|qM) exten = 0033,2,Hangup ngrep shows as follows; U 192.168.0.103:5060 - 192.168.0.3:5070 INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bKa854c86267e80f96..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3 fa865dcc1..To: sip:[EMAIL PROTECTED]:5070..Contact: sip:[EMAIL PROTECTED] 3..Supported: replaces..Call-ID: [EMAIL PROTECTED]: 589 86 INVITE..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Type: ap plication/sdp..Content-Length: 354v=0..o=0303 8000 8000 IN IP4 192.168. 0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m=audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=r tpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap:2 G726-32/8000..a=rtpmap :15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mode=20..a=rtpmap:9 G722/16 000..a=ptime:20.. # U 192.168.0.3:5070 - 192.168.0.103:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 192.168.0.103;b ranch=z9hG4bKa854c86267e80f96;received=192.168.0.103..From: sip:[EMAIL PROTECTED] 68.0.3:5070;tag=c7a5ee3fa865dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01 593a47..Call-ID: [EMAIL PROTECTED]: 58986 INVITE..User-A gent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCR IBE, NOTIFY..Contact: sip:[EMAIL PROTECTED]:5070..Proxy-Authenticate: Dige st algorithm=MD5, realm=asterisk, nonce=72494d6d..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bKa854c86267e80f96..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3fa8 65dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Contact: sip:0303@ 192.168.0.103..Call-ID: [EMAIL PROTECTED]: 58986 ACK..U ser-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,C ANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 INVITE sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;br anch=z9hG4bK6e9ddb4b834276ef..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3 fa865dcc1..To: sip:[EMAIL PROTECTED]:5070..Contact: sip:[EMAIL PROTECTED] 3..Supported: replaces..Proxy-Authorization: Digest username=0303, realm =asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=72494d6 d, response=35378e1d15e71946d8ca187b102d0087..Call-ID: 1c59a92f2174f5ca@ 192.168.0.103..CSeq: 58987 INVITE..User-Agent: Grandstream BT100 1.0.6.8..M ax-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUB SCRIBE..Content-Type: application/sdp..Content-Length: 354v=0..o=0303 8 000 8001 IN IP4 192.168.0.103..s=SIP Call..c=IN IP4 192.168.0.103..t=0 0..m =audio 5004 RTP/AVP 0 8 4 18 2 15 97 9..a=sendrecv..a=rtpmap:0 PCMU/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:4 G723/8000..a=rtpmap:18 G729/8000..a=rtpmap: 2 G726-32/8000..a=rtpmap:15 G728/8000..a=rtpmap:97 iLBC/8000..a=fmtp:97 mod e=20..a=rtpmap:9 G722/16000..a=ptime:20.. # U 192.168.0.3:5070 - 192.168.0.103:5060 SIP/2.0 404 Not Found..Via: SIP/2.0/UDP 192.168.0.103;branch=z9hG4bK6e9ddb4 b834276ef;received=192.168.0.103..From: sip:[EMAIL PROTECTED]:5070;tag=c7a 5ee3fa865dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Call-ID: 1c5 [EMAIL PROTECTED]: 58987 INVITE..User-Agent: Asterisk PBX.. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact : sip:[EMAIL PROTECTED]:5070..Content-Length: 0 # U 192.168.0.103:5060 - 192.168.0.3:5070 ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0..Via: SIP/2.0/UDP 192.168.0.103;branc h=z9hG4bK6e9ddb4b834276ef..From: sip:[EMAIL PROTECTED]:5070;tag=c7a5ee3fa8 65dcc1..To: sip:[EMAIL PROTECTED]:5070;tag=as01593a47..Contact: sip:0303@ 192.168.0.103..Proxy-Authorization: Digest username=0303, realm=asteris k, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=72494d6d, respo nse=9bea041787bf296bcd1c5d730733f615..Call-ID: [EMAIL PROTECTED] .103..CSeq: 58987 ACK..User-Agent: Grandstream BT100 1.0.6.8..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Cont ent-Length: 0 exit 74 received, 0 dropped Regards, Zen ___ --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] meetme-icecast2-ice2
I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz and referenced http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20Ices. But I could not succeed to start ices-2.0.1 as follows; -- Attempting call on Local/[EMAIL PROTECTED] for [EMAIL PROTECTED]:1 (Retry 1) -- Executing Answer(Local/[EMAIL PROTECTED],2, ) in new stack Channel Local/[EMAIL PROTECTED],1 was answered. -- Executing Answer(Local/[EMAIL PROTECTED],1, ) in new stack -- Executing Wait(Local/[EMAIL PROTECTED],1, 1) in new stack -- Executing Wait(Local/[EMAIL PROTECTED],2, 1) in new stack -- Executing MeetMe(Local/[EMAIL PROTECTED],1, 104) in new stack -- Executing ICES(Local/[EMAIL PROTECTED],2, /usr/src/asterisk/contrib/asterisk-ices.xml) in new stack Aug 18 21:54:27 WARNING[5929]: app_ices.c:152 ices_exec: Write failed to pipe: Broken pipe == Spawn extension (stream, 33102, 3) exited non-zero on 'Local/[EMAIL PROTECTED],2' == Spawn extension (stream, 33100, 3) exited non-zero on 'Local/[EMAIL PROTECTED],1' Aug 18 21:54:27 NOTICE[5929]: pbx_spool.c:239 attempt_thread: Call completed to Local/[EMAIL PROTECTED] Which is the correct usage of asteriks-icecast 'icecast-2.2.0 and ices-2.0.1 (ogg)' or 'icecast-2.2.0 and ices-0.4(mp3)'? Regards, Zen ___ 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
[Asterisk-Users] inter-asterisk meetme
Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen ___ 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
Re: [Asterisk-Users] inter-asterisk meetme
William Boehlke wrote : Why do you want to do that? 100 sip users are connected to each CPU(P4 3.0MHz)s, then I would like to broadcast from one sip phone to 500 sip users. If I have 5 microphones in front of me, I can talk to 5 microphones, then 500 users can listen(one-way mode) simultaneously. But that is not elegant. -- Zen -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Zen Kato Sent: Wednesday, August 03, 2005 5:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] inter-asterisk meetme Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.9/62 - Release Date: 8/2/2005 ___ 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 ___ 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
Re: [Asterisk-Users] FreeBSD 5.4 (Asterisk 1.0.9) compile error
Hi, I have also failed the same point. Mine is 5.4-Stable Jul 16, did make world from 5.3 which works * 1.0.6(?) ports and I did cvsup ports-supfile again several minutes ago. NG. -- Zen Darren Wiebe wrote Did you do a make clean? I just, as in 1 hour ago, successfully installed 1.0.9 using the port on FreeBSD. Yeah, even deleted all the files in the asterisk ports , and refreshed it ports collection. Always fails to compile at this point. Am I missing a package dependency somewhere? Hiya, I was just updating Asterisk to 1.0.9 on FreeBSD 5.4, using the new ports updates. The port won't compile I just get this. chan_zap.c: In function `pri_dchannel': chan_zap.c:8391: error: structure has no member named `cause' chan_zap.c:8886: error: structure has no member named `inband_progress' gmake[1]: *** [chan_zap.o] Error 1 gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.0.9/channels' gmake: *** [subdirs] Error 1 *** Error code 2 ___ 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 ___ 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
Re: [Asterisk-Users] Group/Broadcast Voicemail
Hi, I also changed as following sequences; app_voicemail.c 1. Line 3724 tmp[256] to tmp[4096] vm_exec 2. Line 3760 tmp[256] to tmp[4096] append_mailbox 3. Line 3796 tmp[256] to tmp[4096] vm_box_exists 4. Line 3290 tmp[256] to tmp[4096] vm_execmain 5. Line 80 tmp[256] to tmp[4096] #define BASEMAXLINE 6. Line 82 tmp[256] to tmp[4096] #define BASEMAXLINE I tried to copy to 99 mailboxes, but no luck, only could copy to 51 mailboxes. -- Executing VoiceMail(SIP/1021-6bd9, u010302030303040305030603 070308030903100311031203130314031503160317031803190320032103 220323032403250326032703280329033003310332033303340335033603 370338033903400341034203430344034503460347034803490350035103 520353035403550356035703580359036003610362036303640365036603 670368036903700371037203730374037503760377037803790380038103 820383038403850386038703880389039003910392039303940395039603 970398039903) in new stack (snip).. -- User ended message by pressing # -- Playing 'auth-thankyou' (language 'en') Jun 22 17:15:20 NOTICE[11044]: app_voicemail.c:1244 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Jun 22 17:15:25 WARNING[11044]: app.c:994 ast_lock_path: Failed to lock path '': File exists .(snip).. Jun 22 17:15:25 NOTICE[11044]: app_voicemail.c:1244 copy_message: Copying message from [EMAIL PROTECTED] to [EMAIL PROTECTED] Unable to create lock file: No such file or directory I would like to copy to 100-150 mailboxes for one CPU. I also need someone's help. Regards, Zen Kato I did change char tmp[4096], *ext; to 4096 but there's also the same line under vm_execmain but I really don't know anything about programming. I only saw the same line. ___ 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
Re: [Asterisk-Users] Group/Broadcast Voicemail
Hi Robert, Let me guess... mailbox 5103 or 5203 were the last in the list to receive it? Every trials(1-6) I got only 51 mailboxes copied. My quick guess is 256/5(u0103 and xx03s)=51...1, so changing tmp[256] to tmp[4096] does not work. 'Pseudo-diagram' as you mentioned before(6/8/05) is desirable for expandability, but it also did not work. Regards, Zen Kato ___ 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
Re: [Asterisk-Users] Group/Broadcast Voicemail
Hi, Please see inline: In Message-ID: [EMAIL PROTECTED] Robert Goodyear [EMAIL PROTECTED] wrote : On Jun 22, 2005, at 1:50 PM, Zen Kato wrote: Hi Robert, Let me guess... mailbox 5103 or 5203 were the last in the list to receive it? Every trials(1-6) I got only 51 mailboxes copied. My quick guess is 256/5(u0103 and xx03s)=51...1, so changing tmp[256] to tmp[4096] does not work. 'Pseudo-diagram' as you mentioned before(6/8/05) is desirable for expandability, but it also did not work. So what about the variable BASEMAXINLINE? Did you change that and recompile yet? Yes, I changed #define BASEMAXLINE on step 5(line 80) and step 6(line 82) and recomiled each case. Regards, Zen Kato ___ 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
Re: [Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488
I chaged to use from 'rxfax' to 'txfax' and I succeeded to receive the file from * to the FAX under HT488(firmware 1.0.1.2). My OS is 2.6.11-1.27_FC3smp, CVS-v1-0-04/20/05 with ztdummy. spandsp is fun! I made a file: Channel: SIP/4881 MaxRetries: 0 WaitTime: 20 Application: txfax Data: /usr/home/zenkato/voip/asterisk/fax/tif/receive.tif|caller Then I copied this file to /var/spool/asterisk/outgoing. The log is as follows; *CLI -- Attempting call on SIP/4881 for application txfax(/home/zenkato/voip/asterisk/fax/tif/receive.tif|caller) (Retry 1) Channel SIP/4881-baf5 was answered. Lauching txfax(/home/zenkato/voip/asterisk/fax/tif/receive.tif|caller) on SIP/4881-baf5 The remote was made by 'Japan Electric' The remote was made by 'Japan Electric' DIS with final frame tag In state 10 Start tx document CFR with final frame tag In state 4 Start tx page 0 Start tx page 1 MCF with final frame tag In state 14 May 28 12:35:04 NOTICE[13118]: pbx_spool.c:239 attempt_thread: Call completed to SIP/4881 -- I can not find out why 'rxfax' does not work. It might stop after the end of transmitting file from *. My FAX's LCD shows 'transmission error'. I attached the log of 'rxfax'. Is 'ECM(Error Correction Mode)'supported on spandsp-0.0.2pre18? This is Line 81 of the log. Regards, Zen txfax-test Description: Binary data ___ 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
[Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488
Hi, spandsp-0.0.2pre18 works fine for txfax with HT488(version-1.0.1.2), but rxfax doesn't work. After some FAX sounds, it hangup! Could someone tell me how to debug? The following is the * CLI log to 192.168.0.161:43222 -- Executing NoOp(SIP/4881-bde9, ) in new stack -- Executing RxFAX(SIP/4881-bde9, /home/zenkato/voip/asterisk/fax/tif/send22.tif) in new stack Sip read: ACK sip:[EMAIL PROTECTED]:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK3d371a29 From: sip:[EMAIL PROTECTED]:5070;tag=bdf08f36 To: sip:[EMAIL PROTECTED]:5070;tag=as4090e42f Contact: sip:[EMAIL PROTECTED]:43222 Proxy-Authorization: DIGEST username=4881, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=05ea8e51, response=04269c40dad3c4c71a9d53e56ef6c790 Call-ID: [EMAIL PROTECTED] CSeq: 42398 ACK User-Agent: Grandstream HT488 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 12 headers, 0 lines Sip read: BYE sip:[EMAIL PROTECTED]:5070 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK746c2b14 From: sip:[EMAIL PROTECTED]:5070;tag=bdf08f36 To: sip:[EMAIL PROTECTED]:5070;tag=as4090e42f Proxy-Authorization: DIGEST username=4881, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED]:5070, nonce=05ea8e51, response=f18d922cb703582e1f9de0f0e2fd040b Call-ID: [EMAIL PROTECTED] CSeq: 42399 BYE User-Agent: Grandstream HT488 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 11 headers, 0 lines Sending to 192.168.0.161 : 43222 (non-NAT) Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.161:43222;branch=z9hG4bK746c2b14 From: sip:[EMAIL PROTECTED]:5070;tag=bdf08f36 To: sip:[EMAIL PROTECTED]:5070;tag=as4090e42f Call-ID: [EMAIL PROTECTED] CSeq: 42399 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED]:5070 Content-Length: 0 to 192.168.0.161:43222 -- Executing Hangup(SIP/4881-bde9, ) in new stack == Spawn extension (sip, h, 1) exited non-zero on 'SIP/4881-bde9' Destroying call '[EMAIL PROTECTED]' Regards, Zen ___ 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
Re: [Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488
After I did rxfax(...|debug), I got the followings; ..(snip)... DIS: 80 00 ce f4 80 80 81 80 80 80 18 HDLC underflow in state 9 Changed from phase 4 to 3 T4 timeout in state 9 Changed from phase 3 to 4 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DIS: 80 00 ce f4 80 80 81 80 80 80 18 HDLC underflow in state 9 Changed from phase 4 to 3 T2 timeout Start receiving document Changed from phase 3 to 4 DIS: Prefer 256 octet blocks Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm 2D coding Scan line length: 215mm Recording length: Unlimited Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm Minimum scan line time for higher resolutions: T15.4 = T7.7 North American Letter (215.9mm x 279.4mm) North American Legal (215.9mm x 355.6mm) DIS: 80 00 ce f4 80 80 81 80 80 80 18 ..(snip)... Does HT488 version-1.0.1.2 work OK? Zen ___ 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
[Asterisk-Users] can't CLI STOP NOW by zombie MOH
with bad UDP checksum May 16 06:14:14 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: Received packet with bad UDP checksum May 16 06:14:15 NOTICE[5627]: rtp.c:355 ast_rtcp_read: RTP: Received packet with bad UDP checksum Regards, Zen Kato ___ 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
[Asterisk-Users] usb phone(AU-100) and usb phone adapter(TJ560B)
My notebook has three USB ports. I would like to use usb-phone(AU-100) and usb-analogphone-adapter(TJ560B) using 'wcusb','wcfxs' and 'zap/1' and 'zap/2' on CVS-v1-0-03/05/05 on FC3(2.6.11-1.14_FC3). I could not make /dev/zap/1, /dev/zap/2 for usb devices. How should I do? Do I need X100P type(PCI-bus) interface for zap channel for notebook? # lsusb Bus 004 Device 001: ID : Bus 003 Device 002: ID 0d8c:000e C-Media Electronics, Inc. Bus 003 Device 001: ID : Bus 002 Device 003: ID 06e6:c31c Tiger Jet Network, Inc. Bus 002 Device 002: ID 05e3:1205 Genesys Logic, Inc. Afilias Optical Mouse H3003 Bus 002 Device 001: ID : Bus 001 Device 001: ID : # lsmod snd_usb_audio 65153 0 snd_usb_lib13121 1 snd_usb_audio snd_rawmidi28641 1 snd_usb_lib snd_seq_device 8781 1 snd_rawmidi ztdummy 3924 0 wcusb 19616 0 wcfxs 31904 0 zaptel204804 8 ztdummy,wcte11xp,wcusb,wcfxs,wcfxo,wct1xxp,wct4x uhci_hcd 33497 0 ...(snip) # ztcfg ZT_CHANCONFIG failed on channel 2: No such device or address (6) # asterisk -vc .(snip)... [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels .(snip)... # ls -al /dev .(snip).. drwxr-xr-x 2 rootroot 120 4 23 19:02 zap crw--- 1 rootroot196, 253 4 23 19:02 zaptel crw-rw-rw- 1 rootroot 1, 5 4 24 2005 zero # ls -al /dev/zap drwxr-xr-x 2 root root 120 4 23 19:02 . drwxr-xr-x 11 root root 6240 4 23 19:27 .. crw-rw 1 root root 196, 254 4 23 19:02 channel crw-rw 1 root root 196, 0 4 23 19:02 ctl crw-rw 1 root root 196, 255 4 23 19:02 pseudo crw-rw 1 root root 196, 253 4 23 19:02 timer Regards, Zen ___ 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
[Asterisk-Users] gnophone and sip phone
Hi, I installed 'gnophone' on my notePC(RHL9 linux-2.4.20-30.9) with asterisk CVS-02/05/04. I have three unsolved problems: (1)call from gnophone to sip phone is OK, but gnophone's speaker volume is very low even though setting highest volume with gmix, the speaker volume is very high. The sip hardphone side: my voice returns back to earphone of handset(echo?). (2)can not make a call from sip hardphone to gnophone *CLI says as follows; Mar 6 11:27:55 NOTICE[81926]: chan_iax.c:4098 socket_read: Rejected connect attempt from 192.168.0.11, request '[EMAIL PROTECTED]' does not exist Urgent handler Mar 6 11:27:55 WARNING[81926]: chan_iax.c:3951 socket_read: Call rejected by 192.168.0.11: No such context/extension Mar 6 11:27:55 NOTICE[81926]: chan_iax.c:1050 iax_destroy: Avoiding IAX destroy deadlock -- Called [EMAIL PROTECTED] Urgent handler -- Nobody picked up in 5000 ms -- Hungup 'IAX[192.168.0.11:5036]/7' Mar 6 11:28:00 WARNING[294929]: channel.c:517 ast_channel_free: Channel 'IAX[192.168.0.11:5036]/7' may not have been hung up properly Urgent handler Mar 6 11:28:10 WARNING[294929]: pbx.c:1834 ast_pbx_run: Timeout, but no rule 't' in context 'sip' - end of *CLI - my iax.conf is; [916] type=friend host=dynamic defaultip=192.168.0.11 port=5036 secret=916 context=default my extensions.conf is; [default] .. exten = 916,1,Dial(IAX/[EMAIL PROTECTED],5,r) . What is the meaning of '[EMAIL PROTECTED]' above *CLI? Do I miss something in 'iax.conf'? (3)When I start 'gnophone', I have to do the following sequence; 1.start mpg123 some.mp3 2.start 'asterisk' 3.stop mpg123 4.start 'gnophone' Because, asterisk graps sound device and the others can not use sound device after asterisk started. How can I release 'sound device' after asterisk started? Zen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, Thank you for the information. There are ts in Dial command in extensions.conf. When I deleted these ts, each sip phones were directly communicating. I just wrote these ts from the examples. Does these t and T are used for transfer(blind/consaltation) from called user and calling user, respectively? If we don't have these 't' and 'T', can't we do transfer? Regards, Zen Girish Gopinath [EMAIL PROTECTED] wrote : Zen, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? Do you have a Dial statement that has t or T in it? This will force the media stream to pass through Asterisk. Regards, Girish _ Contact brides grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] does usb-ohci work for ztdummy?
Hi, One of my CPU boards is MS-5169(ATX AL9 mainboard). The chipset is Aladdin M1531/M1543. The USB is 'usb-ohci'. ztdummy.c and ztdummy.h uses 'usb-uhci', so I changed from 'uhci' to 'ohci' in ztdummy.c and ztdummy.h. and tried /sbin/modprobe ztdummy, never succeeded. Is it impossible to use 'usb-ohci' instead of 'usb-uhci' for ztdummy? -- Zen Kato ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? -- Zen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
