Re: [asterisk-users] Conferencing..
Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? What do you mean when you describe cards as having or not having Zaptel? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Patch for call deflection with libpri
Hi! Anyone got a patch for call deflection for Zaptel/libpri drivers? Thanks! /hanna ___ -- 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] Conferencing..
Hi Ajey, which kind of BRI are you using? Giorgio Incantalupo Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- 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
Re: [asterisk-users] Conferencing..
You can do conferencing without the zap interface. just modprobe ztdummy. Its good for small conferences. On Mon, 2008-04-14 at 23:39 -0500, Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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 odbc storage
Hi, I was able to store voicemail following the tutorial http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage i would just like to inquire how can i create a web interface (will use php) to play the voicemail stored in the database. the field in the database is recording longblob anyone able to retrieve that file and play on an embedded player on a webpage? Thank You Regards Nhadie ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Please keep us updated on your progress. I am considering putting several of these boxes in and I would love to hear how this comes out. Wish I had something to suggest. Ex Vito wrote: Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Thanks in advance! System: HP Proliant DL380 G5 with 2x PCI-X + 1x PCIe riser card OS: Centos 5 Kernel: 2.6.18-53.1.14.el5 (also tested under 2.6.18-53.el5) HW: Digium TE220B, the one with HW echo cancellation (configured as 2x E1 via jumpers) Context: Pre-site installation of system, no E1 conectivity (loopbacks tested) /etc/zaptel.conf: span=1,1,0,ccs,hdb3,crc4 bchan=25-39,41-55 dchan=40 span=2,2,0,ccs,hdb3,crc4 bchan=56-70,72-86 dchan=71 Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel buffer: About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#0! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042621c] release_console_sem+0x17e/0x1b8 [c0407406] do_IRQ+0xa5/0xae [f8994311] t4_dacs+0x211/0x24b [wct4xxp] [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel] [c0457600] mempool_alloc+0x28/0xc9 [c04ddd33] cfq_resort_rr_list+0x23/0x8b [c04deb6c] cfq_add_crq_rb+0xba/0xc3 [c04dec72] cfq_insert_request+0x42/0x498 [c04d5175] elv_insert+0x10a/0x1ad [c04d908b] __make_request+0x31d/0x366 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04dde27] __cfq_slice_expired+0x8c/0xa5 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04d505d] elv_next_request+0x15c/0x16a [f88bc101] start_io+0x77/0xdc [cciss] [f88bf63e] do_cciss_request+0x32c/0x337 [cciss] [f88ccff0] __split_bio+0x408/0x418 [dm_mod] [f88cd6a6] dm_request+0xce/0xd4 [dm_mod] [c04d6a81] generic_make_request+0x248/0x258 [c04d8734] submit_bio+0xbf/0xc5 [c04548e2] find_get_page+0x18/0x38 [c04719ad] __find_get_block_slow+0xfb/0x105 [c0471cea] __find_get_block+0x15c/0x166 [c0471cea] __find_get_block+0x15c/0x166 [c0471d24] __getblk+0x30/0x270 [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd] [f885a472] journal_cancel_revoke+0x77/0x96 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c041f871] __wake_up+0x2a/0x3d [f8856679] journal_stop+0x1b0/0x1ba [jbd] [c042a209] current_fs_time+0x4a/0x55 [c048626d] touch_atime+0x60/0x8f [c04552ee] do_generic_mapping_read+0x421/0x468 [c045478b] file_read_actor+0x0/0xd1 [c04548e2] find_get_page+0x18/0x38 [c0457319] filemap_nopage+0x192/0x315 [c046048f] __handle_mm_fault+0x85e/0x87b [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Secondary Sync Source Completed startup! Soft lockup ?! Hmmm... I'm ignorant on this, but it smells fishy ! For completeness sake, driver was previously loaded ok: Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 ACPI: PCI Interrupt :18:08.0[A] - GSI 19 (level, low) - IRQ 98 Found TE2XXP at base address fdff, remapped to f8854000 TE2XXP version c01a016a, burst ON Octasic optimized! FALC version: 0005, Board ID: 00 Reg 0: 0x375a2400 Reg 1: 0x375a2000 Reg 2: 0x Reg 3: 0x Reg 4: 0x3101 Reg 5: 0x Reg 6: 0xc01a016a Reg 7: 0x1300 Reg 8: 0x Reg 9: 0x00ff2031 Reg 10: 0x004a TE2XXP: Launching card: 0 TE2XXP: Setting up global serial parameters Found a Wildcard: Wildcard TE220 (4th Gen) After trying lot's of things (disable ILO, disable USBs, try different kernel, different TE220B, etc), I figured that this soft hangup does not show under zaptel 1.4.9.2... In all due honesty, I haven't got the faintest idea what kind of impact this could have. Side testing zaptel 1.4.10 on a simpler system, an HP Proliant ML110 (nearly a PC), the error does not show up as well. I checked the zaptel 1.4.10 ChangeLog
Re: [asterisk-users] Is Asterisk really good??
Quote We had a master source location.with a master image We cloned the hard drive with linux dd copy of master image Did the dd to clone it actually work on RAID devices Mike Trest - On Travel wrote: -Original Message- I'd be interested in sections like Rolling out a new server or How we maintain all the little configuration files without losing our sanity. Hi, I will contribute my 2-cents on how I maintained consistency on a large application with 64 + Asterisks that all had to have the same config and links back to a central DB. Whenever we needed a new machine, we just We had a master source location.with a master image We cloned the hard drive with linux dd copy of master image boot the new machine with this disk assign appropriate IP address perform some sanity checks prior to shipping Send either disk or full machine to remote COLO for physical install. After the machine came on line, it would have enough configuration to join the other members of the farm of asterisks. For intermediate updates, we used SSL-DSA keys between the master master image machine and each of the 64+ remotes. We would wrote our own script and gave it a list of each machine on which to perform the particular steps. When it was launched, we just went out to lunch or home at night while the remotes were updated. This application had as many as 6,000 simultaneous call running and we wrote the scripts such that each remote were placed in a take no calls status by the script so we did not kill any active traffic. We found that no canned package was useful to do this because each maintenance cycle was addressing a different part of the overall configuration and had slightly different commands that were needed. Any good script writer can do the same for what you described. Regards, ..mike.. ___ -- 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
Re: [asterisk-users] Realtime MOH
Pete, Have a look at: http://bugs.digium.com/view.php?id=11196 - Original Message - From: Pete Kay To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, April 01, 2008 12:31 PM Subject: [asterisk-users] Realtime MOH Hi all, I want to allow different users to have their own unique MOH. Is there anyway to do it? Asterisk does not have a realtime MOH feature but I am wondering if there is anyway to get around it? Thank you for your suggestion. Thanks, Pete -- ___ -- 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
Re: [asterisk-users] Problem with SPA3000 -- dropping calls
On 15/04/2008, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all I have SPA3000 (in Linksys reincarnation) and it has very annoying problem. Sometimes, incoming PSTN call drops the moment one picks up analog phone on FXO port. Most of the times it works, other times phone on FXS rings, I pick it up and all I get is a dial tone. Any ideas what may be wrong? You'll need to start by saying where in the world you are, and what Regional settings you have changed from default on the SPA unit. Generally you can google for settings, Country name and SPA in order to find someone who knows how to set up a device for your locale. Regards, Steve ___ -- 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 response 480 Do Not Disturb
Hello, I have switched on DND on a SNOM 360. When I call this phone, I get the following output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8, SIP/user3|20|tr) in new stack -- Called user3 -- Got SIP response 480 Do Not Disturb back from 192.168.0.34 -- SIP/user3-081f8d20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/user4-0821b0e8, fehler|s-CONGESTION|1) in new stack -- Goto (fehler,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/user4-0821b0e8, xCONGESTION) in new stack -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/user4-0821b0e8, ) in new stack I'm using a separate context to catch the dialstatus [fehler] exten = s-NOANSWER,1,NoOp(xNOANSWER) exten = s-NOANSWER,2,Hangup exten = s-CHANUNAVAIL,1,NoOp(xCHANUNAVAIL) exten = s-CHANUNAVAIL,2,Hangup exten = s-BUSY,1,NoOp(xBUSY) exten = s-BUSY,2,Hangup exten = s-CONGESTION,1,NoOp(xCONGESTION) exten = s-CONGESTION,2,Hangup exten = _s-.,1,NoOp() exten = _s-.,2,Hangup Now my question is: Is it possible to tell asterisk that SIP 480 shouldn't result in dialstatus CONGESTION, but in dialstatus BUSY? Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ -- 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.conf wont load completely
14 apr 2008 kl. 16.19 skrev Al lists: I have seen this issue on both 1.2 and 1.4, was not able to reproduce to find a cause or bug. I have seen this after power failure boot up. show sip peer command shows most of peers, except one or two (in my cases trunk) . if i issue a sip reload command, it will show all of them. I can write a script to reload asterisk after a minute of boot up but i wanted to see if anyone else has seen this issue or has any thoughts. Could be a DNS issue. /O ___ -- 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] question about queue
Two use-cases where autofill=no is desirable: 1) If it's important that you answer your callers in strict order (i.e. in order to meet estimated wait time commitments etc). 2) If your queue members/agents are local channels (as local channels are always available, so call attempts will be made regardless of who's talking). Kind regards, Matt. BJ wrote / This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. // // That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. // // BJ/ ___ -- 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 response 480 Do Not Disturb
15 apr 2008 kl. 12.06 skrev Stefan Guenther: Hello, I have switched on DND on a SNOM 360. When I call this phone, I get the following output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8, SIP/user3|20|tr) in new stack -- Called user3 -- Got SIP response 480 Do Not Disturb back from 192.168.0.34 -- SIP/user3-081f8d20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/user4-0821b0e8, fehler|s-CONGESTION|1) in new stack -- Goto (fehler,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/user4-0821b0e8, xCONGESTION) in new stack -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/user4-0821b0e8, ) in new stack I'm using a separate context to catch the dialstatus [fehler] exten = s-NOANSWER,1,NoOp(xNOANSWER) exten = s-NOANSWER,2,Hangup exten = s-CHANUNAVAIL,1,NoOp(xCHANUNAVAIL) exten = s-CHANUNAVAIL,2,Hangup exten = s-BUSY,1,NoOp(xBUSY) exten = s-BUSY,2,Hangup exten = s-CONGESTION,1,NoOp(xCONGESTION) exten = s-CONGESTION,2,Hangup exten = _s-.,1,NoOp() exten = _s-.,2,Hangup Now my question is: Is it possible to tell asterisk that SIP 480 shouldn't result in dialstatus CONGESTION, but in dialstatus BUSY? Well, 480 translates to AST_CAUSE_NOANSWER - cause 19 - check by checking HANGUPCAUSE instead of DIALSTATUS and you will get many more details. /Olle ___ -- 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] Conferencing..
On Mon, 14 Apr 2008, Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. It can do conferencing without a zap interface too. The BRI cards I have they do not have Zaptel. And? How do I enable conferencing on my server? Well you could start by reading the manual, or books on the subject. There's a very good one avalable free for download too. Hint: You're looking for the MeetMe application. You need to read the book; Asterisk the future of telephony. Google for it, you can get it as a PDF. Which will save (as someone else here mentioned recently!) using up newbie karma points... Gordon ___ -- 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] question about queue
With regard to (1), yes, very good point there and certainly reason enough to leave it alone. I had completely forgotten about a use case like that. With regard to (2), I'm pretty sure there's been work done in the recent past to make chan_local more state aware so that this might not be the case any more depending on what version you are using. I might be wrong there, but I know I've got a patch or two hanging around that did make this work. Matt King wrote: Two use-cases where autofill=no is desirable: 1) If it's important that you answer your callers in strict order (i.e. in order to meet estimated wait time commitments etc). 2) If your queue members/agents are local channels (as local channels are always available, so call attempts will be made regardless of who's talking). Kind regards, Matt. BJ wrote / This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. // // That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. // // BJ/ -- -- Bird's The Word Technologies, Inc. http://www.btwtech.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
[asterisk-users] Polycom phone reboots
We are using Asterisk and SER with Polycom 550 phones running SIP version 2.2.2.0084. The phones register to SER. If an AOR appears on more than one phone when a call arrives for that AOR one, some or all of the Polycom phones reboot. I can't seem to find the source of this problem. Has anyone else encountered this problem? Thanks,Steve ___ -- 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 response 480 Do Not Disturb
Johansson Olle E schreef: 15 apr 2008 kl. 12.06 skrev Stefan Guenther: Hello, I have switched on DND on a SNOM 360. When I call this phone, I get the following output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8, SIP/user3|20|tr) in new stack -- Called user3 -- Got SIP response 480 Do Not Disturb back from 192.168.0.34 -- SIP/user3-081f8d20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/user4-0821b0e8, fehler|s-CONGESTION|1) in new stack -- Goto (fehler,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/user4-0821b0e8, xCONGESTION) in new stack -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/user4-0821b0e8, ) in new stack I'm using a separate context to catch the dialstatus [fehler] exten = s-NOANSWER,1,NoOp(xNOANSWER) exten = s-NOANSWER,2,Hangup exten = s-CHANUNAVAIL,1,NoOp(xCHANUNAVAIL) exten = s-CHANUNAVAIL,2,Hangup exten = s-BUSY,1,NoOp(xBUSY) exten = s-BUSY,2,Hangup exten = s-CONGESTION,1,NoOp(xCONGESTION) exten = s-CONGESTION,2,Hangup exten = _s-.,1,NoOp() exten = _s-.,2,Hangup Now my question is: Is it possible to tell asterisk that SIP 480 shouldn't result in dialstatus CONGESTION, but in dialstatus BUSY? Well, 480 translates to AST_CAUSE_NOANSWER - cause 19 - check by checking HANGUPCAUSE instead of DIALSTATUS and you will get many more details. Wouldn't it be more intuitive to translate to AST_CAUSE_BUSY? Ron /Olle ___ -- 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 -- NeoNova BV, The Netherlands Professional internet and VoIP solutions http://www.neonova.nl Kruislaan 419 1098 VA Amsterdam info: 020-5628292 servicedesk: 020-5628292 fax: 020-5628291 KvK Amsterdam 34151241 The following disclaimer applies to this email: http://www.neonova.nl/maildisclaimer ___ -- 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] What kind of Specs for Conference server
Hi List, I know that this question has been asked before so please forgive me. I have a client that wants a box with 24 extensions that will have on it 6 conference rooms. All the phones will be using uLaw. Thanks. Dovid___ -- 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] Conferencing..
Hi Faraz, yes, you can use ztdummy but it cannot completely replace Digium cards. It depends from your hardwareI had troubles with some kind of serversso beware. Giorgio. Faraz R. Khan wrote: You can do conferencing without the zap interface. just modprobe ztdummy. Its good for small conferences. On Mon, 2008-04-14 at 23:39 -0500, Ajey Gore wrote: I figured that asterisk can do conferencing if we have zap interface. The BRI cards I have they do not have Zaptel. How do I enable conferencing on my server? Regards Ajey ___ -- 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 -- _ Giorgio Incantalupo, mailto:[EMAIL PROTECTED] FGA srl - http://www.fgasoftware.com - [EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172 ___ -- 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] Lypp/37 Signals mashup contest
I didn't see any mention of this contest on the mailing list - is anyone using Lypp with their asterisk server for anything funky? http://deancollinsblog.blogspot.com/2008/04/lypp37-signals-mashup-contes t-or-why-i.html http://blog.lypp.com/2008/04/14/37signals-voip-mashup-with-lypp/ Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ___ -- 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] question about queue
Hey, I just found out today that it doesn't work on Asterisk 1.4.19 (at least for realtime queues) if you have autofill=yes in queues.conf. However it works if you add it in queue settings for each queue, for realtime that would be ALTER TABLE queue_table ADD COLUMN autofill TINYINT(1) UNSIGNED DEFAULT 1; For following this issue, see http://bugs.digium.com/view.php?id=12445 Regards, Atis On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ -- 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 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] problem with Asterisk 1.4.19 - accountcode dissapearing
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS CORRECT AND IS
EQUALS TO THE ACCOUNTCODE SPECIFIED MUCH EARLIER IN THE DIALPLAN
exten = _X!.,n,Gotoif($[${i} = 1]?$[${PRIORITY}+2])
;DIAL ALL MAC PHONE ASSOCIATED WITH THIS EXTENSION SIMULATENOUSLY
exten =
_X!.,n,Dial(${mac_dial_string:0:$[${LEN(${mac_dial_string})}-20]}|${sip_phon
es_ring_time}) ;remove least 7 characters, thos
e are left there by the invalid last SQL fetch
exten = _X!.,n,Set(i=0)
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS EMPTY, and so
is this variable if I use it in any way.
When I dial an extension and it hits this diaplan, it works fine. But if I
dial an extension, answer and then transfer (using Polycom phones) to an
extension using this dialplan I lose the accountcode where specified in the
code. It's empty. How can ${CDR(accountcode)} lose it's value for no
reason in those two seemingly innocent diaplan lines?
Below is the CLI output if it's useful:
-- Executing [EMAIL PROTECTED]:22]
NoOp(SIP/0004f2134384-1-097fb4e8, 1234567890) in new stack ;THIS IS THE
ACCOUNTCODE
-- Executing [EMAIL PROTECTED]:23]
GotoIf(SIP/0004f2134384-1-097fb4e8, 0?25) in new stack
-- Executing [EMAIL PROTECTED]:24]
Dial(SIP/0004f2134384-1-097fb4e8, SIP/0004f2134384-3|8) in new stack
-- Called 0004f2134384-3
-- SIP/0004f2134384-3-099947b0 is ringing
== Spawn extension (generic-extensions-db, 705, 24) exited non-zero on
'SIP/0004f2134384-1-097fb4e8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back from
192.168.1.6
-- Nobody picked up in 8000 ms
-- Executing [EMAIL PROTECTED]:25]
Set(SIP/0004f212ae63-1-099700a8, i=0) in new stack
-- Executing [EMAIL PROTECTED]:26]
NoOp(SIP/0004f212ae63-1-099700a8, ) in new stack ;MISSING ACCOUNTCODE
IS HERE
Mick
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Re: [asterisk-users] Zap Codec
But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Monday, April 14, 2008 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Monday, April 14, 2008 14:39 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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] problem with Asterisk 1.4.19 - accountcode dissapearing
As far as I noticed - this issue is not 1.4.19 only. Same thing happens on
all Asterisk versions.
Set your own variable before transfer:
Exten = , Set(__MYACC=${CDR(accountcode)})
And use ${MYACC} in other (transfered) calls.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Tuesday, April 15, 2008 3:24 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] problem with Asterisk 1.4.19 - accountcode
dissapearing
Hi,
I have a big issue during transfers (using Polycom phones, but I don't think
that's relevent) with Asterisk 1.14.19. Basically, the value contained in
${CDR(accountcode)} dissapears.
Here is the relevant code snippet:
--
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS CORRECT AND IS
EQUALS TO THE ACCOUNTCODE SPECIFIED MUCH EARLIER IN THE DIALPLAN
exten = _X!.,n,Gotoif($[${i} = 1]?$[${PRIORITY}+2])
;DIAL ALL MAC PHONE ASSOCIATED WITH THIS EXTENSION SIMULATENOUSLY
exten =
_X!.,n,Dial(${mac_dial_string:0:$[${LEN(${mac_dial_string})}-20]}|${sip_phon
es_ring_time}) ;remove least 7 characters, thos
e are left there by the invalid last SQL fetch
exten = _X!.,n,Set(i=0)
exten = _X!.,n,Noop(${CDR(accountcode)}) ;THE VALUE HERE IS EMPTY, and so
is this variable if I use it in any way.
When I dial an extension and it hits this diaplan, it works fine. But if I
dial an extension, answer and then transfer (using Polycom phones) to an
extension using this dialplan I lose the accountcode where specified in the
code. It's empty. How can ${CDR(accountcode)} lose it's value for no
reason in those two seemingly innocent diaplan lines?
Below is the CLI output if it's useful:
-- Executing [EMAIL PROTECTED]:22]
NoOp(SIP/0004f2134384-1-097fb4e8, 1234567890) in new stack ;THIS IS THE
ACCOUNTCODE
-- Executing [EMAIL PROTECTED]:23]
GotoIf(SIP/0004f2134384-1-097fb4e8, 0?25) in new stack
-- Executing [EMAIL PROTECTED]:24]
Dial(SIP/0004f2134384-1-097fb4e8, SIP/0004f2134384-3|8) in new stack
-- Called 0004f2134384-3
-- SIP/0004f2134384-3-099947b0 is ringing
== Spawn extension (generic-extensions-db, 705, 24) exited non-zero on
'SIP/0004f2134384-1-097fb4e8ZOMBIE'
-- Incoming call: Got SIP response 500 Internal Server Error back from
192.168.1.6
-- Nobody picked up in 8000 ms
-- Executing [EMAIL PROTECTED]:25]
Set(SIP/0004f212ae63-1-099700a8, i=0) in new stack
-- Executing [EMAIL PROTECTED]:26]
NoOp(SIP/0004f212ae63-1-099700a8, ) in new stack ;MISSING ACCOUNTCODE
IS HERE
Mick
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Re: [asterisk-users] Zap Codec
The PSTN only allows ulaw or alaw (depending on your location). You CANNOT send calls in any other codec over a PSTN line. Generally, if you want to use G729 then you must buy a G729 license (with a few exceptions). Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Darryl Dunkin *Sent:* Monday, April 14, 2008 9:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Zap Codec This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jeremy Mann *Sent:* Monday, April 14, 2008 14:39 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I’m not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Zap Codec
So in other words, if I have G729 enabled on the phones, I must get G729 licenses to use Zap channels. Otherwise I have to use ULAW for everything? I fail to understand why it'd be difficult to do codec negotiation on SIP-ZAP calls, Zap sends that it only supports ulaw, if the phone doesn't then the call is cancelled or forwarded to logic to translate. I realize G729 is fairly cheap, but it's useless server overhead when the phone supports the codec it needs natively. Is there any dialplan logic that can coerce the transaction to be ulaw only? Setting something in the SIP header perhaps? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec The PSTN only allows ulaw or alaw (depending on your location). You CANNOT send calls in any other codec over a PSTN line. Generally, if you want to use G729 then you must buy a G729 license (with a few exceptions). Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Darryl Dunkin *Sent:* Monday, April 14, 2008 9:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Zap Codec This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jeremy Mann *Sent:* Monday, April 14, 2008 14:39 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper
Re: [asterisk-users] Problem with SPA3000 -- dropping calls
You need to clarify what you mean by: 1. pick up phone on FXO port (Where is this phone attached? are you branching the incoming PSTN where one goes to SPA and one to a normal phone?) 2. Phone on FXS? the FXS on the SPA itself? How does that work? do you have call routing setup that way in asterisk? On Tue, 2008-04-15 at 10:26 +0100, Steve Davies wrote: On 15/04/2008, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, all I have SPA3000 (in Linksys reincarnation) and it has very annoying problem. Sometimes, incoming PSTN call drops the moment one picks up analog phone on FXO port. Most of the times it works, other times phone on FXS rings, I pick it up and all I get is a dial tone. Any ideas what may be wrong? You'll need to start by saying where in the world you are, and what Regional settings you have changed from default on the SPA unit. Generally you can google for settings, Country name and SPA in order to find someone who knows how to set up a device for your locale. Regards, Steve ___ -- 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 -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.111.111.320 x200 www.emergen.biz ___ -- 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] Zap Codec
If you are talking between two g729 endpoints, the Asterisk overhead is very small. Jeremy Mann wrote: So in other words, if I have G729 enabled on the phones, I must get G729 licenses to use Zap channels. Otherwise I have to use ULAW for everything? I fail to understand why it'd be difficult to do codec negotiation on SIP-ZAP calls, Zap sends that it only supports ulaw, if the phone doesn't then the call is cancelled or forwarded to logic to translate. I realize G729 is fairly cheap, but it's useless server overhead when the phone supports the codec it needs natively. Is there any dialplan logic that can coerce the transaction to be ulaw only? Setting something in the SIP header perhaps? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 8:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec The PSTN only allows ulaw or alaw (depending on your location). You CANNOT send calls in any other codec over a PSTN line. Generally, if you want to use G729 then you must buy a G729 license (with a few exceptions). Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Darryl Dunkin *Sent:* Monday, April 14, 2008 9:01 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Zap Codec This is SIP channel you need to limit. Forcing ulaw only, the Polycom will fall back to ulaw. Per peer, in your sip.conf: disallow=all allow=ulaw *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Jeremy Mann *Sent:* Monday, April 14, 2008 14:39 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Zap Codec Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the
Re: [asterisk-users] What kind of Specs for Conference server
On Tue, 15 Apr 2008, Dovid B wrote: I know that this question has been asked before so please forgive me. Easier to ask for forgiveness than permission? I have a client that wants a box with 24 extensions that will have on it 6 conference rooms. All the phones will be using uLaw. I don't think you can buy a PC so slow it couldn't handle 24 simultaneous calls -- you may be able to find something in your junk pile that wouldn't handle it. Seriously, I tested a $300 Zonbu box (1.2gHz VIA processor, 512mb ram) with 180 calls. (I forget if it was SIP or IAX -- you could search the mailing list archives...) The calls were only playing demo-congrats in a loop, but the audio quality was fine. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- 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 on EC2
I have tried to install an Asterisk server on a CentOS EC2 image. The install went ok. I was able to connect with X-lite to the instance and the instance apparently played back SayDigits(123) (see below) Connected to Asterisk 1.4.19 currently running on domU-12-31-38-00-91-42 (pid = 13114) Verbosity is at least 5CLI -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/xlite1-081b83f0, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/xlite1-081b83f0, 2) in new stack -- Executing [EMAIL PROTECTED]:3] SayDigits(SIP/xlite1-081b83f0, 123) in new stack -- SIP/xlite1-081b83f0 Playing 'digits/1' (language 'en') -- SIP/xlite1-081b83f0 Playing 'digits/2' (language 'en') -- SIP/xlite1-081b83f0 Playing 'digits/3' (language 'en') == Auto fallthrough, channel 'SIP/xlite1-081b83f0' status is 'UNKNOWN' -- Executing [EMAIL PROTECTED]:1] Ringing(SIP/xlite1-081ca050, ) in new stack -- Executing [EMAIL PROTECTED]:2] Wait(SIP/xlite1-081ca050, 2) in new stack -- Executing [EMAIL PROTECTED]:3] SayDigits(SIP/xlite1-081ca050, 123) in new stack -- SIP/xlite1-081ca050 Playing 'digits/1' (language 'en') -- SIP/xlite1-081ca050 Playing 'digits/2' (language 'en') -- SIP/xlite1-081ca050 Playing 'digits/3' (language 'en') == Auto fallthrough, channel 'SIP/xlite1-081ca050' status is 'UNKNOWN' However, I DO NOT hear the actual playback on X-lite. I feel that something is wrong with non-opened ports. Can anyone help? Did anyone succeed in getting Asterik to run on EC2? Philippe ___ -- 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] gotoif syntax error
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = GROUP,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} : in=([^|]+)]
Always]?r,1)
The error is for the last line (IN,1). Funny thing is that asterisk doesnt
report any error for the first line (OUT,1)
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Re: [asterisk-users] Unable to load module chan_zap.so
On 15/04/2008, at 10:34 PM, Tzafrir Cohen wrote:
On Tue, Apr 15, 2008 at 10:33:43AM +0800, Jeremy Malcolm wrote:
On 14/04/2008, at 5:58 PM, Tzafrir Cohen wrote:
In the Asterisk CLI, what happens when you run:
This is Asterisk 1.2:
unload chan_zap.so
load chan_zap.so
Yeah, I mentioned in my first post to the list that that doesn't
work. It just says Unable to load module chan_zap.so but doesn't
say why, no matter how verbose I set it to be. It's the same message
as if I had typed load nonexistent_module.so.
--
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Internet and Open Source lawyer, IT consultant, actor
host -t NAPTR 1.0.8.0.3.1.2.9.8.1.6.e164.org|awk -F! '{print $3}'
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Re: [asterisk-users] Zap Codec
On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Tilghman ___ -- 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] Zap Codec
That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 Sys Admin in Chicago IL (full time)
If this is posted in the wrong place, I apologize profusely in advance. Please advise me as to the correct place to post it! Ifbyphone Inc. is a fully funded startup located in Skokie, IL (Chicago suburb) that provides a platform of Software as a Service functions to bridge the web to the telephone. We are currently searching for an energetic Junior Systems Administrator to act as a backup to the Telecommunications Manager and also to perform some basic scripting development as time permits. Our ideal candidate would be located in the greater Chicago area, have a minimum of 1 year of professional experience in the workplace, and would posses at least a larger portion of the following skills: Systems Administration -- CentOS / RHEL 4.x Apache 2.x Postfix / Sendmail PHP 4/5 MySQL 5 IPtables SNMP Win2k3 Standard Hardware Raid Basic WinXP and Mac OSX desktop support Networking -- Basic Cisco SNMP MRTG Basic VPN (IPSEC/PPTP) Basic BGP QoS Basic VLAN T1/PRI Asterisk -- 1.4.x (NO Trixbox/FreePBX/PBX in a Flash/etc) Dialplan config SIP/IAX2/RTP AGI/FastAGI Call Files General -- Basic PHP/MySQL development Soft Skills -- Can work independently Can complete projects by a deadline with minimal supervision Able to keep up with rapidly changing priorities Willing to carry pager or cellphone for on-call rotation These items are *not* an absolute must-have. A candidate that has a larger portion of these skills, and can learn quickly is also of interest to us! Relocation assistance is an option for the right candidate. Please submit your resume via email to [EMAIL PROTECTED] or fax to 847-676-6553 for immediate consideration. ___ -- 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] Zap Codec
Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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] polycom 501 stopped working
Hi all, I have a polycom 501 phone that I rebooted today. It stopped working... Normally the screen shows New call, Forward and that is all... Now the screen shows New call, Forward, MyStat, Buddies. It no longer accepts incoming calls nor can I make outgoing calls. I have reloaded factory defaults, and rebooted, reset and rebooted and it wont go back to normal... What happened and how do I get it back to normal? All my phones boot my TFTP, This is working as I see the requests in the log file. I was playing with sip.cfg (which I never had used at all before). I have now removed the sip.cfg I created and rebooted the phone but it still isnt working... The sip.cfg file is the only file I created/changed and now I have removed it. What gives? Jerry ___ -- 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] [VOIP-Users-Conference] Re: Free FAX license from Pika
Randy, I think you are simplifying the issue by saying it's because of asterisk that they are no longer required. They also had a very important use when sending bulk international faxes in that they provided store and forward functionality. Eg you send a fax to japan but you would fax to a local usa number, they would then transmit via a private IP network and then drop it out the other end via a local call in Tokyo. Yes I know it's hard for people to think that a company like that could be doing 10's of millions of dollars in traffic a month but you have to remember a lot of things have changed. Keep this in mind when thinking that Asterisk is the be all and end all to voip.sure as the sun rises tomorrow there will be something to replace it OR it must morph into something else. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). -Original Message- From: [EMAIL PROTECTED] [mailto:VOIP-Users- [EMAIL PROTECTED] On Behalf Of randulo Sent: Tuesday, 15 April 2008 10:45 AM To: [EMAIL PROTECTED] Subject: [VOIP-Users-Conference] Re: Free FAX license from Pika On Tue, Apr 15, 2008 at 4:36 PM, Darren [EMAIL PROTECTED] wrote: No I don't remember Jfax. For the benefit of you younger members, Jfax (aka J2, jconnect) http://home.j2.com is one of many services that most asterisk users don't need anymore :) Something like long distance callback services. I used to use these when they were a way to save money. That time is long gone. /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
Re: [asterisk-users] gotoif syntax error
מוישי ברעוודה wrote:
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = GROUP,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
in=([^|]+)] Always]?r,1)
The error is for the last line (IN,1). Funny thing is that asterisk
doesnt report any error for the first line (OUT,1)
Because OUT is correct. IN is missing a =, as in = Always
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Re: [asterisk-users] SIP response 480 Do Not Disturb
15 apr 2008 kl. 13.38 skrev Ron Arts: Johansson Olle E schreef: 15 apr 2008 kl. 12.06 skrev Stefan Guenther: Hello, I have switched on DND on a SNOM 360. When I call this phone, I get the following output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8, SIP/user3|20|tr) in new stack -- Called user3 -- Got SIP response 480 Do Not Disturb back from 192.168.0.34 -- SIP/user3-081f8d20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/user4-0821b0e8, fehler|s-CONGESTION|1) in new stack -- Goto (fehler,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/user4-0821b0e8, xCONGESTION) in new stack -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/user4-0821b0e8, ) in new stack I'm using a separate context to catch the dialstatus [fehler] exten = s-NOANSWER,1,NoOp(xNOANSWER) exten = s-NOANSWER,2,Hangup exten = s-CHANUNAVAIL,1,NoOp(xCHANUNAVAIL) exten = s-CHANUNAVAIL,2,Hangup exten = s-BUSY,1,NoOp(xBUSY) exten = s-BUSY,2,Hangup exten = s-CONGESTION,1,NoOp(xCONGESTION) exten = s-CONGESTION,2,Hangup exten = _s-.,1,NoOp() exten = _s-.,2,Hangup Now my question is: Is it possible to tell asterisk that SIP 480 shouldn't result in dialstatus CONGESTION, but in dialstatus BUSY? Well, 480 translates to AST_CAUSE_NOANSWER - cause 19 - check by checking HANGUPCAUSE instead of DIALSTATUS and you will get many more details. Wouldn't it be more intuitive to translate to AST_CAUSE_BUSY? Well, we're following the IETF standards, talk with them :-) I guess SNOM should have a setting so that the phone actually sends BUSY if you want it to send BUSY. /O ___ -- 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] gotoif syntax error
I may of removed it for testing just prior to send the email - i get the
same error when its there:
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} : in=([^|]+)]
= Always]?r,1)
On Tue, Apr 15, 2008 at 6:07 PM, Jason Parker [EMAIL PROTECTED] wrote:
מוישי ברעוודה wrote:
Asterisk is reporting the following error:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax
error: syntax error, unexpected ':', expecting $end; Input:
: Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = GROUP,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
in=([^|]+)] Always]?r,1)
The error is for the last line (IN,1). Funny thing is that asterisk
doesnt report any error for the first line (OUT,1)
Because OUT is correct. IN is missing a =, as in = Always
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ipconnect, ltd.
26 Strauss St., Jerusalem, Israel
W. 1.800.800.456 (+9722.569.5295)
M. +97254.666.1367
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Re: [asterisk-users] polycom 501 stopped working
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset the local config for the phone On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis [EMAIL PROTECTED] wrote: Hi all, I have a polycom 501 phone that I rebooted today. It stopped working... Normally the screen shows New call, Forward and that is all... Now the screen shows New call, Forward, MyStat, Buddies. It no longer accepts incoming calls nor can I make outgoing calls. I have reloaded factory defaults, and rebooted, reset and rebooted and it wont go back to normal... What happened and how do I get it back to normal? All my phones boot my TFTP, This is working as I see the requests in the log file. I was playing with sip.cfg (which I never had used at all before). I have now removed the sip.cfg I created and rebooted the phone but it still isnt working... The sip.cfg file is the only file I created/changed and now I have removed it. What gives? Jerry ___ -- 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 -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 ___ -- 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] Zap Codec
If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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] Zap Codec
I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] polycom 501 stopped working
try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset the local config for the phone On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi all, // // I have a polycom 501 phone that I rebooted today. It stopped working... // Normally the screen shows New call, Forward and that is all... // Now the screen shows New call, Forward, MyStat, Buddies. // It no longer accepts incoming calls nor can I make outgoing calls. // // I have reloaded factory defaults, and rebooted, reset and rebooted // and it wont go back to normal... // // What happened and how do I get it back to normal? // // All my phones boot my TFTP, This is working as I see the requests // in the log file. // // I was playing with sip.cfg (which I never had used at all before). // I have now removed the sip.cfg I created and rebooted the phone // but it still isnt working... // // The sip.cfg file is the only file I created/changed and now I have // removed it. // // What gives? // // Jerry/ Still coming up with the Mystat menu after doing that... Jerry ___ -- 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] question about queue
Two use-cases where autofill=no is desirable: 1) If it's important that you answer your callers in strict order (i.e. in order to meet estimated wait time commitments etc). Not always the case. Let's look at multiple queue assignment where agents have skills (logged in) to multiple queues. AGT1: Has SkillA, SkillB, SkillC AGT2: Has SKillA SLA: 24 seconds Senario Calls in Queues: Call1: SkillA - 15 seconds Call2: SkillB - 12 seconds AGT1 will become available in now() +2 seconds AGT2 will become available in now() +6 seconds CASE 1 (Calls in strict order): TIME=now()+2: AGT1 becomes available, CALL1 matched, time in Q now 17 seconds, assigned, SLA OK. TIME=now()+6: AGT2 becomes available, CALL2 NOT matched, not assigned, AGT2 idle, awaiting AGT1 to finish call, time in Q now 18 seconds. TIME=now()+10: AGT2 idle, CALL2 sitting in queue, SLA failed. CASE 2 (Calls not in order, system SMART enough to read into the queue and predict availability based on historical data) TIME=now()+2: AGT1 becomes available, CALL1 matches, but system knows that CALL2 is also a match and remaining agents are NOT a match. Predicted availability says call 2 will fail SLA, system assigns CALL2 to AGT2, time in Q now 14 seconds, SLA OK. TIME=now()+6: AGT2 becomes available, CALL1 matches and is assigned, time in Q now 21 seconds, SLA OK. TIME=now()+10: AGT1 on call 2, SLA OK. AGT2 on call 1, SLA OK. Now this isn't strictly the problem originally described but I'm trying to articulate where the use case as specified falls down in real-world environments. This also shows and area that Asterisk (and _many_ other switches) have not gone yet but we need to aspire to. This type of functionality is why you currently shell out the bucks for Avaya. - dbc. ___ -- 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] PBX Console
Originally posted by: mailto: Hi, I've been looking into the one bad thing about * which is there's no practical solution to running a console. You know the kind where you have rows of buttons each representing an extension. You press the button of the extension you want to transfer the call to, and it's done. There's the beginnig of GUI version but it's going to eat resources for running X which can become less than desirable, besides it's not very competitive having to use a mouse to handle calls. Too slow. So my idea is to have a text window. We can run at a higher res than 25x80 and squeeze a fair number of extensions onto it. The idea is to either use the extension number to access an extension or for less than 100 station system, use a two digit number for each person. This way there's minimum typing for the operator. This have enough space to easily display busy, hold, vmail etc. as the status of each extension. This way with a flatscreen monitor, or dual for bigger systems we can even run the console away from the server and use minimum bandwidth. The other status screen would be a voice mail screen where you can A) see the status of voicemail. Lines in use etc. B) change the name and features associated with voice mail. -- Steve Szmidt HTML in e-mail is not safe. It let's spammers know to spam you more, and sets you up for online attack through IE 4.x and above. Using HTML in e-mail only promotes it as safe to the uninitiated. ___ Asterisk-Users mailing list mailto:Asterisk-Users@lists.digium.com 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
Re: [asterisk-users] polycom 501 stopped working
You mentioned that you removed sip.cfg. Do you have the file with the way it was before you made any changes? If so, try placing it back where it's supposed to be, then reboot the server, then reboot the phone. (p.s. I don't have an *, I don't know anything about your problem! this is just suggestions, based upon what you have written, please and don't yell at me if it doesn't work. :) es Jerry Geis wrote: try rebotting the phone. at boot up. hold down 4,8,6,* - that should reset the local config for the phone On Tue, Apr 15, 2008 at 5:58 PM, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Hi all, // // I have a polycom 501 phone that I rebooted today. It stopped working... // Normally the screen shows New call, Forward and that is all... // Now the screen shows New call, Forward, MyStat, Buddies. // It no longer accepts incoming calls nor can I make outgoing calls. // // I have reloaded factory defaults, and rebooted, reset and rebooted // and it wont go back to normal... // // What happened and how do I get it back to normal? // // All my phones boot my TFTP, This is working as I see the requests // in the log file. // // I was playing with sip.cfg (which I never had used at all before). // I have now removed the sip.cfg I created and rebooted the phone // but it still isnt working... // // The sip.cfg file is the only file I created/changed and now I have // removed it. // // What gives? // // Jerry/ Still coming up with the Mystat menu after doing that... Jerry ___ -- 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
[asterisk-users] dialed number notify at invalid dial situation
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback(' is incorrect! please again ')
# This exten lines are figure for instruction.
# I know to use with gsm filename.
but ${EXTEN} meaning 'i' that isn't dialed number.
Does anyone have good idea?
please help
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan.
http://www.dairiten.com:81/modules/news/
powered by xoops at http://www.xoops.org
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[asterisk-users] asterisk online indicator
Originally posted by: mailto: Hi all a funny tool for your * server and portal site. http://www.dairiten.com:81/asterisk_online/indicator.php enjoy :-) # please notify me if this icon design has problem. --- Masakazu Nakano. dairiten.com - an VoIP and Ubiquitus Portal site in Japan. http://www.dairiten.com:81/modules/news/ powered by xoops at http://www.xoops.org ___ Asterisk-Users mailing list mailto:Asterisk-Users@lists.digium.com 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
[asterisk-users] app_swift v1.6.1 released for Asterisk 1.6
Thought I'd let everyone know I've released app_swift v1.6.1 which is entirely based off of Will Orton's work he's placed in the public domain. Works great with Asterisk v1.6.0-beta7.1. In any case, can be downloaded from my site at: http://www.darrensessions.com Go easy on me, this is my first release of anything. Thanks, - Darren ___ -- 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] Problem with SIP, attended transfer and GROUP_COUNT
Hi,
maybe someone can give me a hint to solve the following issue. I want to
limit the calls to a specific SIP-destination. Disabling callwaiting at
the phones is not an option, because it should be configured via the *
database.
My solution uses GROUP_COUNT, which works fine most of the time. In case
of attended transfer (on SIP-basis, not via the #-mechanism of asterisk)
I have problems.
To simplfy the scenario I stripped down the dialplan to the following.
From somewhere on the wiki I am using the following context:
exten = 200,1,Set(GROUP()=${CALLERID(num)})
exten = 200,n,GotoIf($[${GROUP_COUNT(${EXTEN})} = 1]?BLOCK)
exten = 200,n,Set(OUTBOUND_GROUP=${EXTEN})
exten = 200,n,Dial(SIP/katrin)
exten = 200,n(BLOCK),Busy
This block is used for other extensions 100 and 150 respectivily. It
works fine until I am using attended transfer.
Example: kwe (Extension 100) is calling katrin (Extension 200). katrin
sets the call on hold and talks to hans (Extension 150).
At the cli I get the following result:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/katrin-081b70a8200 (default)
SIP/katrin-081bb020200 (default)
SIP/hans-0816b8b8 150 (default)
which seems correct to me.
In case of a transfer of kwe to hans (katrin leaving), the result is:
pbxtest*CLI group show channels
ChannelGroup Category
SIP/kwe-081bf188 100 (default)
SIP/kwe-081bf188 200 (default)
SIP/hans-0816b8b8 150 (default)
I am confused about the second line, which leads to trouble. The above
context would think, that katrin is busy. In case of a blind transfer
everything is ok (the second line does not exist)
I have tested the above with * 1.4.14, 1.4.18-rc4 and 1.4.19
Is this a bug or a feature? Am I doing something wrong or should I file
a bug report?
Thanks in advance,
Regards
Karsten
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Re: [asterisk-users] app_swift v1.6.1 released for Asterisk 1.6
What is app_swift ? Zoa Darren Sessions wrote: Thought I'd let everyone know I've released app_swift v1.6.1 which is entirely based off of Will Orton's work he's placed in the public domain. Works great with Asterisk v1.6.0-beta7.1. In any case, can be downloaded from my site at: http://www.darrensessions.com Go easy on me, this is my first release of anything. Thanks, - Darren ___ -- 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
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Ex Vito, [comments inline] Ex Vito wrote: Hi list, After a lot of testing + troubleshooting, I guess I'm observing what I am now calling a regression with zaptel 1.4.10 (is it?) As such I call for peer feedback, before either asking Digium install support or filing a bug. Under zaptel 1.4.10, when ztcfg runs this gets logged in the kernel buffer: About to enter spanconfig! Done with spanconfig! About to enter spanconfig! Done with spanconfig! About to enter startup! TE2XXP: Span 1 configured for CCS/HDB3/CRC4 timing source auto card 0! wct2xxp: Setting yellow alarm on span 1 timing source auto card 0! SPAN 1: Primary Sync Source VPM400: Not Present VPM450: echo cancellation for 64 channels BUG: soft lockup detected on CPU#0! [c044d448] softlockup_tick+0x96/0xa4 [c042ddc8] update_process_times+0x39/0x5c [c04196f7] smp_apic_timer_interrupt+0x5b/0x6c [c04059bf] apic_timer_interrupt+0x1f/0x24 [f89bc1e7] init_vpm450m+0x32d/0x34a [wct4xxp] [f89a3b11] t4_vpm450_init+0x18ce/0x198c [wct4xxp] [f89a7ee4] t4_startup+0x4315/0x43c7 [wct4xxp] [c042621c] release_console_sem+0x17e/0x1b8 [c0407406] do_IRQ+0xa5/0xae [f8994311] t4_dacs+0x211/0x24b [wct4xxp] [f8a01f6a] zt_ioctl+0x273/0x144f [zaptel] [c0457600] mempool_alloc+0x28/0xc9 [c04ddd33] cfq_resort_rr_list+0x23/0x8b [c04deb6c] cfq_add_crq_rb+0xba/0xc3 [c04dec72] cfq_insert_request+0x42/0x498 [c04d5175] elv_insert+0x10a/0x1ad [c04d908b] __make_request+0x31d/0x366 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04dde27] __cfq_slice_expired+0x8c/0xa5 [c04de8b1] cfq_dispatch_requests+0x26a/0x46b [c04d505d] elv_next_request+0x15c/0x16a [f88bc101] start_io+0x77/0xdc [cciss] [f88bf63e] do_cciss_request+0x32c/0x337 [cciss] [f88ccff0] __split_bio+0x408/0x418 [dm_mod] [f88cd6a6] dm_request+0xce/0xd4 [dm_mod] [c04d6a81] generic_make_request+0x248/0x258 [c04d8734] submit_bio+0xbf/0xc5 [c04548e2] find_get_page+0x18/0x38 [c04719ad] __find_get_block_slow+0xfb/0x105 [c0471cea] __find_get_block+0x15c/0x166 [c0471cea] __find_get_block+0x15c/0x166 [c0471d24] __getblk+0x30/0x270 [f885a485] journal_cancel_revoke+0x8a/0x96 [jbd] [f885a472] journal_cancel_revoke+0x77/0x96 [jbd] [f885626f] __journal_file_buffer+0x10e/0x1e3 [jbd] [c041f871] __wake_up+0x2a/0x3d [f8856679] journal_stop+0x1b0/0x1ba [jbd] [c042a209] current_fs_time+0x4a/0x55 [c048626d] touch_atime+0x60/0x8f [c04552ee] do_generic_mapping_read+0x421/0x468 [c045478b] file_read_actor+0x0/0xd1 [c04548e2] find_get_page+0x18/0x38 [c0457319] filemap_nopage+0x192/0x315 [c046048f] __handle_mm_fault+0x85e/0x87b [c047f46b] do_ioctl+0x47/0x5d [c047f6cb] vfs_ioctl+0x24a/0x25c [c047f725] sys_ioctl+0x48/0x5f [c0404eff] syscall_call+0x7/0xb === VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 2 span(s) Completed startup! About to enter startup! TE2XXP: Span 2 configured for CCS/HDB3/CRC4 wct2xxp: Setting yellow alarm on span 2 timing source auto card 0! SPAN 2: Secondary Sync Source Completed startup! Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set Cheers, Shaun ___ -- 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] Zap Codec
Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on. If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are
Re: [asterisk-users] dialed number notify at invalid dial situation
you could try to set a var to the exten maybe.. and then use that var ..
since when in exten = i , well i will be the exten..
On Tue, Apr 15, 2008 at 11:52 AM, Anonymous [EMAIL PROTECTED] wrote:
Originally posted by: mailto:
Hi all
Now I'm making IVR sequance that is customised [mainmanu].
I wish to notify invaid command like a following
exten = i,1,playback('your command is ...')
exten = i,2,playback(${EXTEN}) ; Say 'i' oops! ;-(
exten = i,3,playback(' is incorrect! please again ')
# This exten lines are figure for instruction.
# I know to use with gsm filename.
but ${EXTEN} meaning 'i' that isn't dialed number.
Does anyone have good idea?
please help
---
Masakazu Nakano.
Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan.
http://www.dairiten.com:81/modules/news/
powered by xoops at http://www.xoops.org
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Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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Re: [asterisk-users] PBX Console
FOP works for us, no need for X: http://www.asternic.org If you need to avoid using a mouse, you can use the Polycom attendant console instead: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound point_ip_attendant_console.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anonymous Sent: Tuesday, April 15, 2008 08:55 To: asterisk-users@lists.digium.com Subject: [asterisk-users] PBX Console Originally posted by: mailto: Hi, I've been looking into the one bad thing about * which is there's no practical solution to running a console. You know the kind where you have rows of buttons each representing an extension. You press the button of the extension you want to transfer the call to, and it's done. There's the beginnig of GUI version but it's going to eat resources for running X which can become less than desirable, besides it's not very competitive having to use a mouse to handle calls. Too slow. So my idea is to have a text window. We can run at a higher res than 25x80 and squeeze a fair number of extensions onto it. The idea is to either use the extension number to access an extension or for less than 100 station system, use a two digit number for each person. This way there's minimum typing for the operator. This have enough space to easily display busy, hold, vmail etc. as the status of each extension. This way with a flatscreen monitor, or dual for bigger systems we can even run the console away from the server and use minimum bandwidth. The other status screen would be a voice mail screen where you can A) see the status of voicemail. Lines in use etc. B) change the name and features associated with voice mail. -- Steve Szmidt HTML in e-mail is not safe. It let's spammers know to spam you more, and sets you up for online attack through IE 4.x and above. Using HTML in e-mail only promotes it as safe to the uninitiated. ___ Asterisk-Users mailing list mailto:Asterisk-Users@lists.digium.com 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 ___ -- 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] Zap Codec
Correct, but if I have two sip peers, one with G729ulaw, the other with gsmulaw, they will negotiate before trying to send audio. With ZAP, it tries to transcode whatever it receives into ulaw, period. No negotiation to even tell the client to send ulaw if capable. With no call level control(or dialplan logic, or anything!), I either use ulaw for ALL CALLS from sip peers(to other sip peers, to iax peers, to ZAP peers/channels), or use a combination of codecs and make sure it's able to be transcoded for the ZAP channels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Tuesday, April 15, 2008 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on. If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] What kind of Specs for Conference server
Dovid B wrote: Hi List, I know that this question has been asked before so please forgive me. I have a client that wants a box with 24 extensions that will have on it 6 conference rooms. All the phones will be using uLaw. And how many total users on the system, including outside conferencing parties? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Zap Codec
Correct, those are two peers talking direct, one call leg (SIP-SIP). In this case, you have two call legs which are then bridged: SIP - Asterisk Asterisk - Zap You've already negotiated g729 before Asterisk notices that the call is going out Zap (via your dialplan). At this point, you have to transcode if your peer is set to use g729. Otherwise, force your SIP end to talk ulaw. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 11:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec Correct, but if I have two sip peers, one with G729ulaw, the other with gsmulaw, they will negotiate before trying to send audio. With ZAP, it tries to transcode whatever it receives into ulaw, period. No negotiation to even tell the client to send ulaw if capable. With no call level control(or dialplan logic, or anything!), I either use ulaw for ALL CALLS from sip peers(to other sip peers, to iax peers, to ZAP peers/channels), or use a combination of codecs and make sure it's able to be transcoded for the ZAP channels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin Sent: Tuesday, April 15, 2008 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec Asterisk builds two channels and bridges them together. If the codecs mis-match then it must transcode, the negotiation on the Zap end is done seperately from the SIP end, so it does not care what your handset decided on. If you want ulaw, use ulaw, not g729 (on any call leg). You won't be able to mix and match codecs between calls, choose one for all calls and stick with it. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Mann Sent: Tuesday, April 15, 2008 08:24 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec I guess that's my frustration, I don't want it g729, I want it ulaw, I just wish Zap did codec negotiation from the client. It'd be a nice option instead of automatically trying to translate if it's not ulaw. Could save some processor overhead(obviously at the expense of bandwidth). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec If you want to get a G729 call to go via Zap you must purchase a G729 license. No amount of discussion is going to change that. Jeremy Mann wrote: Sadly you are correct: -- Executing [EMAIL PROTECTED]:4] Set(SIP/156-083514c0, _SIP_CODEC=ulaw) in new stack -- Executing [EMAIL PROTECTED]:5] NoOp(SIP/156-083514c0, 4) in new stack -- Executing [EMAIL PROTECTED]:6] NoOp(SIP/156-083514c0, ) in new stack -- Executing [EMAIL PROTECTED]:7] Dial(SIP/156-083514c0, ) in new stack [Apr 15 09:51:25] WARNING[17434]: channel.c:3265 ast_request: No translator path exists for channel type Zap (native 76) to 256 [Apr 15 09:51:25] WARNING[17434]: app_dial.c:1191 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [EMAIL PROTECTED]:8] Hangup(SIP/156-083514c0, ) in new stack -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Tuesday, April 15, 2008 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Zap Codec That would work just spiffy if you are calling another SIP device, but by the time the call gets to that point in the dialplan the codec of the originating device has already been chosen and set in stone. Tilghman Lesher wrote: On Tuesday 15 April 2008 07:40:47 Jeremy Mann wrote: But I want my polycom to attempt g729 on SIPPEER-SIPPEER calls, I only want ulaw used when SIPPEER-ZAP is the case. Set(_SIP_CODEC=ulaw) Dial(Zap/g0/...) -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- 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 This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any
Re: [asterisk-users] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set ...thanks for your feedback Shaun. I am currently nearing other troubleshooting issues regarding a TC400B (which will probably lead me to get in touch with Digium install support). So I have no schedule today to test your suggestions; maybe tomorrow / thursday. They are noted, however. :) Cheers, -- exvito ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set Opps, forgot to feedback: yes this kernel seems to have CONFIG_4KSTACKS enabled. -- exvito ___ -- 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] app_swift v1.6.1 released for Asterisk 1.6
An app to invoke the Cepstral text-to-speech engine. On Tue, Apr 15, 2008 at 11:46 AM, Zoa [EMAIL PROTECTED] wrote: What is app_swift ? Zoa ___ -- 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] app_swift v1.6.1 released for Asterisk 1.6
Kai-Uwe Jensen wrote: An app to invoke the Cepstral text-to-speech engine. On Tue, Apr 15, 2008 at 11:46 AM, Zoa [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: What is app_swift ? Zoa I've written an AGI wrapper for it as well, in case you don't want to re-compile to support. http://www.voip-info.org/wiki/view/DTSwift+Cepstral+AGI+Wrapper -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
exvito - I know it is a pain in the cahoonkus - but would you consider sharing the OTHER Digium board issues you are having , the recommended steps you were given by Digium to troubleshoot them, and the results ? I think this real-wold experience wold be invaluable to the list. THX in Advance for sharing ! Ex Vito wrote: Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set ...thanks for your feedback Shaun. I am currently nearing other troubleshooting issues regarding a TC400B (which will probably lead me to get in touch with Digium install support). So I have no schedule today to test your suggestions; maybe tomorrow / thursday. They are noted, however. :) Cheers, -- exvito ___ -- 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
Re: [asterisk-users] PBX Console
Darryl Dunkin wrote: FOP works for us, no need for X: http://www.asternic.org If you need to avoid using a mouse, you can use the Polycom attendant console instead: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound point_ip_attendant_console.html We recently released Maestro Control Panel (beta): http://www.datatrakpos.com/pos/datatalk/maestro.aspx Its mouse driven, but easy to use. We have a few clients using it for their secretaries with good success. You can minimize it to the system tray and it'll popup when flagged numbers come in or click on it to do things like get so and so on the phone for me type of functions. It uses the manager api for its functionality so its pretty flexible. We're also working on a cross platform (Win/Linux) sister product designed to run on small touch screens systems. The idea is that it will run on a small embedded linux box (maybe fastened to the underneath of the desk) using a small 8 touch screen. Nothing to show yet unfortunately. You can check back into the message board every so often for news of it when its released. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
Shaun - Could you clarify your post a bit ? 1 - Is the 4 K stacks a Known Problem ? a) If so is it known to be problem on any specific Linux distro ? b) Should ALL installation Check for this PRIOR to doing an Asterisk Install ? 2) The branch you mention below - are fixes from it in Any current * release ? Shaun Ruffell wrote: Your stack trace appears to possibly be stack corruption. Could you try either this branch: http://svn.digium.com/view/zaptel/team/mattf/zaptel-1.4-stackcleanup/ Or with a kernel that does not have 4K stacks enabled? You can check if your installed kernel does with the following command. $ cat /boot/config-`uname -r` | grep 4K # CONFIG_4KSTACKS is not set Cheers, Shaun ___ -- 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
Re: [asterisk-users] voicemail odbc storage
On Tue, 2008-04-15 at 16:22 +0800, nhadie ramos wrote: i would just like to inquire how can i create a web interface (will use php) to play the voicemail stored in the database. This really isn't the proper venue for that type of question... but searching Google for PHP BLOB returns a large number of hits, and several of the ones I looked at showed very good instructions on how to get a BLOB from the database via PHP code. the field in the database is recording longblob anyone able to retrieve that file and play on an embedded player on a webpage? I've seen it done many times before. The secret here is to find a good embedded media player and make sure the audio file is in the correct format. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- 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] gotoif syntax error
actually I screw up a lot as i changed something for testing. here is the
correct error/dialplan:
[Apr 15 16:58:32] WARNING[14759] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected '=', expecting $end; Input:
= Always
^
here is the dialplan:
exten = OUT,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = GROUP,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} :
out=([^|]+)] = Always]?r,1)
exten = IN,1,Gotoif($[$[${DB(AMPUSER/${ARG1}/recording)} : in=([^|]+)]
= Always]?r,1)
The error is for the last line (IN,1). Funny thing is that asterisk doesnt
report any error for the first line (OUT,1)
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Re: [asterisk-users] Analog DID
It seems the standard for Analog DID (at least around here) is wink start, does the Rhino cards work with this or do I need to have the telco immediately send the DTMF tones? On Wed, Feb 13, 2008 at 12:33 PM, James Finstrom [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rhino's Analog cards support analog DID. no need for all the extra stuff You will want to get an R8FXX with fxs modules that will give you channels in sets of 2. ADID has not really taken off in the OS telephony market I think due to a lack of understanding people stay with the proprietary phone systems that pimp this feature. Okay so I will take the lead and pimp it for asterisk. With Rhino Analog cards you CAN do ADID with no extra equipment. However if you want to spend the money we can go the other route :) darren wrote: An analog DID trunk is a line (typically part of a group) that has a group of numbers assigned to it at the telco side. They work in a variety of ways depending on the telco. One example is the trunks as Telus provides them. The end user provides dialtone back to the telco. When a call comes in on a DID the telco picks up the first available line (remember, the customer is providing dial tone.) and dials the last 4 digits of the dialed number. They are often replaced by PRIs but in some locations a PRI is not affordable and these provide the same DID functionality for a small fraction of the price. Darren Wiebe [EMAIL PROTECTED] Wed Feb 13 2008 10:11:44 AM MST from Tzafrir Cohen to asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Analog DID On Wed, Feb 13, 2008 at 10:40:25AM -0600, Joe Pukepail wrote: Does anyone have any suggestions for connecting analog DID trunks? What is an analog DID trunk? You want to connect phones to your Asterisk? Connect to the PSTN? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4 Port FXS module for channel bank - $150 T1 Card - $500 This is for providing plenty of analog extensions (phones). Is that what you're after? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/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 !DSPAM:47b327c1163231152562594! -- ___ -- 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 !DSPAM:47b327c1163231152562594! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD4DBQFHsynrdloC7YyaIOoRAuKhAJiCRxUX+E7rzt6/A5nyAjXdO5yaAJ4/HoKB Gxd6H7YOdzXfygVuBygzAw== =51QY -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
Re: [asterisk-users] SIP response 480 Do Not Disturb
The only solution that I found for this is to use Asterisk 1.4 with devstate backport (http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/) and use the hints and to determine if it's inuse (or any other status) before the dialing - in order to generate a proper reply. I didn't find a way to handle the SIP 480 reply using Asterisk 1.2 properly. Note that it's an idea I was about to run but I didn't get to it yet. devstate on test machine compiled fine seems to be working from first sight. Tomer. Stefan Guenther wrote: Hello, I have switched on DND on a SNOM 360. When I call this phone, I get the following output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8, SIP/user3|20|tr) in new stack -- Called user3 -- Got SIP response 480 Do Not Disturb back from 192.168.0.34 -- SIP/user3-081f8d20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/user4-0821b0e8, fehler|s-CONGESTION|1) in new stack -- Goto (fehler,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/user4-0821b0e8, xCONGESTION) in new stack -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/user4-0821b0e8, ) in new stack I'm using a separate context to catch the dialstatus [fehler] exten = s-NOANSWER,1,NoOp(xNOANSWER) exten = s-NOANSWER,2,Hangup exten = s-CHANUNAVAIL,1,NoOp(xCHANUNAVAIL) exten = s-CHANUNAVAIL,2,Hangup exten = s-BUSY,1,NoOp(xBUSY) exten = s-BUSY,2,Hangup exten = s-CONGESTION,1,NoOp(xCONGESTION) exten = s-CONGESTION,2,Hangup exten = _s-.,1,NoOp() exten = _s-.,2,Hangup Now my question is: Is it possible to tell asterisk that SIP 480 shouldn't result in dialstatus CONGESTION, but in dialstatus BUSY? Thanks for your help, Stefan ___ -- 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] CDR and transfers! :(
Hi list, I've been reading the archives and I know that transfers are unimplemented in the CDR ( http://lists.digium.com/pipermail/asterisk-users/2007-June/189902.html) In my case I'm running Asterisk 1.4.17 as an Office PBX. In this setup just a small group of users are able to make long distance call directly from his/her extension, so the ones who can't needs to ask the operator/receptionist to call these numbers and then transfer (almost always atxfer) the call back to them. In this case, I want to account the whole call duration to the user that have requested it, not the operator, but I cannot manage to find a way to map the CDR generated by the operator with the CDR generated by the transfer. This is an example of the generated CDRs by atxfer: John Doe make a long distance call, talk a few seconds and then make an attended transfer to extension 128. John Doe 111,111,128,SIP/128-0071d340,2008-04-15 14:31:31,2008-04-15 14:31:32,2008-04-15 14:31:41,10,9,ANSWERED,1208286068.2256 John Doe 111,111,90212555,Zap/4-1,2008-04-15 14:31:08,2008-04-15 14:31:14,2008-04-15 14:31:41,33,27,ANSWERED,1208286068.2255 111,111,s,,2008-04-15 14:31:08,2008-04-15 14:31:14,2008-04-15 14:31:41,33,27,ANSWERED,1208286068.2256 111,111,s,Zap/4-1,2008-04-15 14:31:31,2008-04-15 14:31:32,2008-04-15 14:31:47,16,15,ANSWERED,1208286091.2260 This is an example of the generated CDRs by blind xfer: John Doe make a long distance call, talk a few seconds and then make a blind transfer to extension 128. John Doe 111,111,904124172996,Zap/6-1,2008-04-11 13:10:48,2008-04-11 13:10:55,2008-04-11 13:11:13,25,18,ANSWERED,1207935648.5256 111,111,128,SIP/128-006fec80,2008-04-11 13:11:13,2008-04-11 13:11:22,2008-04-11 13:11:32,19,10,ANSWERED,1207935673.5258 In this case I think you can guess the uniqueid of the transferred call by adding the call duration (25 in this example) made by John Doe to the uniqueid of the same call, but that's not the case in the atxfer scenario. I think this is a very common scenario so, how are you doing to handle this situation??? -- Raul Gomez Linux Counter #156439 ___ -- 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] PBX Console
What is the default username/password. In the Maestro forum's it only says it's hardcoded, but doesn't say the actual username/password. Best Regards, On Tue, Apr 15, 2008 at 4:43 PM, Lee Jenkins [EMAIL PROTECTED] wrote: Darryl Dunkin wrote: FOP works for us, no need for X: http://www.asternic.org If you need to avoid using a mouse, you can use the Polycom attendant console instead: http://www.polycom.com/usa/en/products/voice/desktop/soundpoint_ip/sound point_ip_attendant_console.html We recently released Maestro Control Panel (beta): http://www.datatrakpos.com/pos/datatalk/maestro.aspx Its mouse driven, but easy to use. We have a few clients using it for their secretaries with good success. You can minimize it to the system tray and it'll popup when flagged numbers come in or click on it to do things like get so and so on the phone for me type of functions. It uses the manager api for its functionality so its pretty flexible. We're also working on a cross platform (Win/Linux) sister product designed to run on small touch screens systems. The idea is that it will run on a small embedded linux box (maybe fastened to the underneath of the desk) using a small 8 touch screen. Nothing to show yet unfortunately. You can check back into the message board every so often for news of it when its released. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- 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 -- Guilherme Loch Góes Visite nossa loja virtual: http://www.shopvoip.com.br Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br ___ -- 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] Global call limit
Hi, i'm new in asterisk programming. Maybe my question was posted thousand times but i found nothing using google. I'm looking for a method to limit the total simultaneous calls (inbound and outbound) that pass from internal phones to 2 SIP providers. I found the calllimit option but it works only on a per-channel basis. Instead i want limit the total amount of calls, abstracting from which SIP provider us used. This to keep good audio quality for active calls and rejecting new arriving: this is needed for a PBX connectect with a poor ADSL having only 256kbit in upload: so i want permit only 3 calls (256 / 80 kbit) and rejecting the 4th. Any solutions? Thanks. -- PicoStreamer - the real WEB live streaming software vinz486.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
[asterisk-users] Good article about VoIP, etc.
Gang, I know some of you like to keep up-to-date on various VoIP-ish happenings. Here's an interesting little article about FreeSWITCH that also mentions Asterisk: http://digg.com/software/Freeswitch_Poised_to_Shake_Up_the_Open_Source_V oIP_Scene The author guesstimates that Asterisk has roughly 95% of the OSS telephony market. I'd be interested to know if anyone has hard facts about the market share that Asterisk/Digium enjoy, both from the OSS telephony perspective as well as compared to the big commercial vendors. -MC ___ -- 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] Zap Codec
In the sip peer definition, disallow=all allow=g729 allow=ulaw SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw for the ZAP calls. But, when your polycoms talk with each other, as long as all necessary REINVITEs happen, they should use the 729 codec I believe. Remember however, that many options to the Dial application, like t,w,m,k (or so) REQURE asterisk to remain in the media path. moj Jeremy Mann wrote: Is there a way to force Zap channels to only use ulaw, and not even attempt g729 negotiation? My polycom 550 has G729,ulaw as priority, the Zap always fails b/c I'm not licensed for the codec on the asterisk box. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. ___ -- 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 -- *Mojo Wentworth* HORAN COMPANY, LLC 403 Lincoln Street, Suite 210 Sitka, AK 99835 (907) 747- (907) 747-7417 - Fax [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
Re: [asterisk-users] Global call limit
On Wed, Apr 16, 2008 at 12:46 AM, Vinz486 [EMAIL PROTECTED] wrote:
Hi,
i'm new in asterisk programming.
Maybe my question was posted thousand times but i found nothing using google.
I'm looking for a method to limit the total simultaneous calls
(inbound and outbound) that pass from internal phones to 2 SIP
providers.
I found the calllimit option but it works only on a per-channel basis.
Instead i want limit the total amount of calls, abstracting from which
SIP provider us used.
This to keep good audio quality for active calls and rejecting new
arriving: this is needed for a PBX connectect with a poor ADSL having
only 256kbit in upload: so i want permit only 3 calls (256 / 80 kbit)
and rejecting the 4th.
Any solutions?
if (${GROUP_COUNT([EMAIL PROTECTED])}) function in combination with
Set(GROUP(a)=x)
or Set([EMAIL PROTECTED])
Regards,
Atis
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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Re: [asterisk-users] compilation of asterisk 1.4.19 with ilbc already on system
Vieri wrote: How can I tell the make system in 1.4.19 that ilbc is already on the system and that it should link to /usr/lib/libilbc.a? Shouldn't the configure script do that? No; the Asterisk build system has never had support for using a system-provided version of the iLBC library. Whoever provided you that library could easily run afoul of the same licensing issues that caused us to remove the code from our Asterisk distribution, and using that library does not obviate you from the need to register your intent to use the codec if you are using it for commercial purposes. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] PBX Console
Guilherme Loch Waltrick Góes wrote: What is the default username/password. In the Maestro forum's it only says it's hardcoded, but doesn't say the actual username/password. Guilherme, The username is leebo and the password is 123. You can see it by going to: Admin Users Edit Users and selecting my record :) == Lee Jenkins. Are you having trouble entering the software? If so, it's a good chance you may be running Vista or a nicely locked down version of XP. The original installer saved Maestro's (firebirdsql) database file to the (\program files\Maestro Control Panel) directory which was a mistake on my part. Windows may be refusing to let you connect to the database because of that. I have changed the installer to save these files to CSIDL_COMMON_DOCUMENTS (Users\All Users) folder so there is no longer any problems accessing the database, assuming you have that problem. Either way, I recommend you download it again directly from here: http://www.datatrakpos.com/pos/datatalk/downloads/maestro_setup.zip That's the same link on the webpage, if you don't feel like navigating back. Please post any further support questions to the message board if its not too much trouble: http://www.leebo.dreamhosters.com/pbxbb/ Thanks for downloading and sorry about the inconvenience. -- Warm Regards, Lee When my company started out, we were really, really, really, really small. Now...we're just really small. ___ -- 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] zaptel 1.4.10 regression with TE220B on Proliant DL380 G5 ?
On Tue, Apr 15, 2008 at 8:37 PM, Al Baker [EMAIL PROTECTED] wrote: exvito - I know it is a pain in the cahoonkus - but would you consider sharing the OTHER Digium board issues you are having , the recommended steps you were given by Digium to troubleshoot them, and the results ? I think this real-wold experience wold be invaluable to the list. THX in Advance for sharing ! ...sure, here it goes, without all the infinite detail we went through in the process. Short version: same DL380 G5 system, Centos 5, kernel 2.6.18-53.1.14.el5, zaptel 1.4.10, zaptel 1.4.9.2, almost all possible combinations in PCI slots, USB / 2nd NIC / ILO enabling / disabling. TC400B module loading fails (wctc4xxp) (actually it loaded fine once or twice and asterisk recognized its presence, but failed in subsequent reboots without any reconfiguration!) If asterisk 1.4.19 is started under these conditions, we get a kernel panic -- did not get a dump / log of it but we have a console picture that we can share (~460KiB). But at some point we get: ... [address] apic_timer_interrupt+0x1f/0x24 [address] zt_tc_open+0x59/0xc3 [zttranscode] [address] zt_open+0x86/0x22a [zaptel] [address] chrdev_open+0x11e/0x123 ... We also tried the same card under all the other variations in a different system -- a proliant ML110 G4 -- we obtained the same behaviour: once or twice it loaded most of the time it failed with the same error. dmesg snippet is: ... Zaptel Version: 1.4.10 Zaptel Echo Canceller: MG2 Zaptel Transcoder support loaded Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) Zaptel DTE (G.729a / G.723.1) Transcoder support LOADED (firm ver = 6.12) wctc4xxp: probe of :0a:01.0 failed with error -5 ... Both when the card is the only one installed on the system and when in the presence of TE220B and / or TE122. We contacted Digium support, who suggested we RMA this card, they believe the card is faulty. We seem to agree, as the behavior does not seem to make much sense (although this is our first experience with such a card) There it is, in the hope that it helps some one in the future. We will post back results when the new card arrives. Cheers, -- exvito ___ -- 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] wcfxo and X100P card won't play nice.
Greetings, This may have already been asked many times, but I cannot seem to find a satisfactory and consistent answer anywhere. I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850 or 2650 (cannot recall): 00:00.0 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) (rev 13) 00:00.1 Host bridge: Broadcom CMIC-WS Host Bridge (GC-LE chipset) 00:00.2 Host bridge: Broadcom CMIC-LE 00:04.0 Class ff00: Dell Embedded Remote Access or ERA/O 00:04.1 Class ff00: Dell Remote Access Card III 00:04.2 Class ff00: Dell Embedded Remote Access: BMC/SMIC device 00:0e.0 VGA compatible controller: ATI Technologies Inc Rage XL (rev 27) 00:0f.0 Host bridge: Broadcom CSB5 South Bridge (rev 93) 00:0f.1 IDE interface: Broadcom CSB5 IDE Controller (rev 93) 00:0f.2 USB Controller: Broadcom OSB4/CSB5 OHCI USB Controller (rev 05) 00:0f.3 ISA bridge: Broadcom CSB5 LPC bridge 00:10.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:10.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.0 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 00:11.2 Host bridge: Broadcom CIOB-X2 PCI-X I/O Bridge (rev 03) 01:06.0 Communication controller: Motorola Wildcard X100P 03:06.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 03:08.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5701 Gigabit Ethernet (rev 15) 04:08.0 PCI bridge: Intel Corporation 80303 I/O Processor PCI-to-PCI Bridge (rev 01) 04:08.1 RAID bus controller: Dell PowerEdge Expandable RAID Controller 3/Di (rev 01) But, wcfxo won't recognise it: NOTICE-wcfxo: WCFXO/0: Unknown DAA chip revision: REVB=0 Failed to initailize DAA, giving up... wcfxo: probe of :01:06.0 failed with error -5 This is running on a custom-compiled kernel 2.6.24.3 with Asterisk 1.4.18 and Zaptel 1.4.10. Any ideas? -- Alex -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Zap Codec
On Tue, Apr 15, 2008 at 11:37 PM, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: In the sip peer definition, disallow=all allow=g729 allow=ulaw SHOULD work. Asterisk can't transcode g729, so it should fall on ulaw for the ZAP calls. But, when your polycoms talk with each other, as long as all necessary REINVITEs happen, they should use the 729 codec I believe. Remember however, that many options to the Dial application, like t,w,m,k (or so) REQURE asterisk to remain in the media path. moj AFAICT, I say that in this case this will not work... Very unfortunatelly. It's related to the way the current asterisk versions behave regarding codec negotiation / renegotiation. Your sip.conf entry will have the phone-asterisk leg be g729 and the other leg, to the PSTN, will be a/u-law. When bridging, asterisk is not clever enough (yet!) to renegotiate the SIP leg back to a/u-law and either a) it transcodes or b) the call fails if no transcoder is available... I've given this issue some testing with no sucessful results in the recent past... (check last two/three months list archives) Asterisk really needs a revamped media renegotiation algorithm ! Will we get one in 1.6 ?!... I guess not. Again, unfortunatelly, as this is a very core, very important issue. (feel free to correct me and give me the good news !!!) Cheers, -- exvito ___ -- 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] X-Lite and Presence?
Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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] question about queue
Yup, I am using realtime queue. Do you mean the global setting in queue.conf is useless and you have to set every thing in each queue to activate the settings? If it is true, does it also apply to other realtime settings? On Tue, Apr 15, 2008 at 8:21 PM, Atis Lezdins [EMAIL PROTECTED] wrote: Hey, I just found out today that it doesn't work on Asterisk 1.4.19 (at least for realtime queues) if you have autofill=yes in queues.conf. However it works if you add it in queue settings for each queue, for realtime that would be ALTER TABLE queue_table ADD COLUMN autofill TINYINT(1) UNSIGNED DEFAULT 1; For following this issue, see http://bugs.digium.com/view.php?id=12445 Regards, Atis On Sat, Apr 12, 2008 at 4:42 AM, Rilawich Ango [EMAIL PROTECTED] wrote: Do you mean autofill works in 1.4.x? But it doesn't work even I set it. On Fri, Apr 11, 2008 at 11:07 AM, BJ Weschke [EMAIL PROTECTED] wrote: Rilawich Ango wrote: Thanks. I have checked that the queue.conf. I keep the default setting as autofill=yes in my tests. That's mean even autofill=yes, the 1st caller will still stick the whole queue. asterisk version : 1.4.18 --queue.conf-- ; AutoFill Behavior ;The old/current behavior of the queue has a serial type behavior ;in that the queue will make all waiting callers wait in the queue ;even if there is more than one available member ready to take ;calls until the head caller is connected with the member they ;were trying to get to. The next waiting caller in line then ;becomes the head caller, and they are then connected with the ;next available member and all available members and waiting callers ;waits while this happens. The new behavior, enabled by setting ;autofill=yes makes sure that when the waiting callers are connecting ;with available members in a parallel fashion until there are ;no more available members or no more waiting callers. This is ;probably more along the lines of how a queue should work and ;in most cases, you will want to enable this behavior. If you ;do not specify or comment out this option, it will default to no ;to keep backward compatibility with the old behavior. ; autofill = yes This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. BJ -- Bird's The Word Technologies, Inc. http://www.btwtech.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 ___ -- 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 -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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
Re: [asterisk-users] X-Lite and Presence?
Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 !DSPAM:48055196261001131342032! ___ -- 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] TXFax/RXFax/AGX-Addons/SpanDSP Crashing
No progress at all. Version from Debian/Lenny repository still crashes and I'm not able to compile AGX. It gives out a long list of error messages. Some unsatisfied dependencies...? I Can't experiment for a while after unwanted night-time visit of fire-fighters :-( I have to let everything dry and clean out of sand and drywall pieces :-( Martin - Original Message - From: Justin Newman To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Sent: 11. dubna 2008 13:00 Subject: Re: [asterisk-users] TXFax/RXFax/AGX-Addons/SpanDSP Crashing Did this just start happening with the 1.4 tree? Have you made any progress on getting it resolved? Justin Newman Matt Riddell wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tzafrir Cohen wrote: Let's be more specific here, folks: What version numbers? Asterisk, spandsp, agx-addons / rx-tx-fax? Asterisk: yesterday's 1.4 SVN SpanDSP: tried with pre 15, 16 and 18 AGX-Addons: tried with 1.4.5 and svn trunk rx/txfax: supplied by AGX Addons - although they seem to build the files and stick them into the modules directory, rather than adding to the apps directory and modifying the Makefile. i have Asterisk 1.4.18, SpanDSP 0.0.4pre16, AGX addons 1.4.5 linux kernel 2.6.18 AMD64. it (Asterisk) segfault on rxfax when i enable faxdetect in zapata.conf. since then it disabled faxdetect and use nvfaxdetect function in dialplan, it works fine afterward. also it seems to works fine using regular 32bit kernel. -- Edwin Lam [EMAIL PROTECTED] Systems Engineer, Office General, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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___ -- 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] X-Lite and Presence?
Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 !DSPAM:48055196261001131342032! ___ -- 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
Re: [asterisk-users] X-Lite and Presence?
X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 ___ -- 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 !DSPAM:48056806261001804221289! ___ -- 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] X-Lite and Presence?
Thanks again!.. Right. I have it working now, it shows the users statuses as online or offline and changes them when someone closes their app. But not free/busy type changes.. Any idea why here? Simon On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote: X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 ___ -- 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 !DSPAM:48056806261001804221289! ___ -- 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
[asterisk-users] [asterisk-announce] Zaptel 1.2.25 and 1.4.10
How does one apply the patch file that is on the site (downloads.digium.com). Thanks J Werh [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
Re: [asterisk-users] X-Lite and Presence?
IIRC Asterisk doesn't support the full presence publishing spec so you won't get the full range of possible status types, however you should at least get free/busy. I vaguely recall having to change the presence type from peer-to-peer to something else - that's done in the SIP configuration window. However, since I don't have X-Lite in front of me at the moment (fortunately, for the most part!) I can't give you more of a hint than that. Simon wrote: Thanks again!.. Right. I have it working now, it shows the users statuses as online or offline and changes them when someone closes their app. But not free/busy type changes.. Any idea why here? Simon On Wed, Apr 16, 2008 at 3:21 PM, Rob Hillis [EMAIL PROTECTED] wrote: X-Lite. Of course, Asterisk will need a hint configured for that extension as well... Simon wrote: Thanks for the reply.. Sorry for the lame question.. Do i do that in X-Lite or Asterisk? On Wed, Apr 16, 2008 at 2:07 PM, Rob Hillis [EMAIL PROTECTED] wrote: Configure the extension as a softphone using the format extension@asterisk.ip.address. Works fine for me - and works even better for agents! Simon wrote: Hi There, We have some users using x-lite as their SIP phone... but im wondering how to get the Calls Contacts to show as being available (Or if it can be done at all?). Is this what Presence is? Thanks Simon ___ -- 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 ___ -- 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 ___ -- 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 !DSPAM:48057d5e261007514015341! ___ -- 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
