[Asterisk-Users] iax2 disconnect problem
Hi, I'm using asterisk 1.2.7.1 and somehow my iax trunking is getting these problem :S. Sometimes iax acts weird and start to drop calls randomly and give these at the log: May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL May 16 13:44:22 DEBUG[5264] chan_iax2.c: Immediately destroying 6, having received INVAL Do I have to post this on mantis? Is this a bug? Anyone can confirm this? Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk hung again
Hi, Yesterday, one of my asterisk servers was hung... On the log, I found these: May 2 09:38:26 DEBUG[28201] rtp.c: RTP Transmission error of packet 50596 to :16480: Network is unreachable May 2 09:38:29 WARNING[17120] chan_sip.c: sip_xmit of 0x81f60b8 (len 396) to :5060 returned -1: Network is unreachable These messages above exist for each devices that are registered to asterisk server. After I restarted the asterisk server, the problem was gone. What did caused this? I'm running asterisk 1.2.7.1 on Redhat EL4 Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma Card Question
Hi Matt, I guess this is the problem within asterisk which wrongly assume the hangup as on-hold call. Do you/your staffs/your customer hang up the phone so quickly that asterisk mistakenly belief that the act is for call waiting? As we know to do some call waiting we just flash the hook swiftly and the other person will hear a music-on-hold. Then if we put the handset down then the phone will ringing once each a couple seconds to remind us that there is call waiting on the phone ;) To avoid this behaviour, try to flash the hook a little longer when hang up the phone (about 2 seconds will be enough).. Regards, Stevanus Matt wrote: By "the system" you mean the phone company? Or asterisk? So what you are saying is I hang up... the sangoma hangups... but the phone company sees it as a flash... then says.. HEY DUDE! YOU JUST HUNG UP ON YOUR CALLER.. Here they are back *ring*. ? On 5/2/06, Melcon Moraes <[EMAIL PROTECTED]> wrote: Maybe some kind of callwaiting/threewaycalling activated on that? The system is identifying the hang up as a flash. -Original Message- From: Matt <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Cc: Sent: Tue, 2 May 2006 16:30:56 -0400 Delivered: Tue, 02 May 2006 17:28:33 Subject:[Asterisk-Users] Sangoma Card Question Hi, I have a Sangoma 200A (I think that's the model #) analog 4 port card. It works great... however almost everytime after someone hangs up a call they were on.. the system rings the call back in, as though it were a new call coming in. When they pickup no one is there. Can anyone suggest why this is happening, and how I can make it stop? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz&_l=1,1146602038.579235.10961.baladonia.terra.com.br,4024,Des15,Des15 --Original Message Ends-- -- Melcon Moraes <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple asterisk process ?
I've tried cat /proc/*asterisk proc number*/environ | strings | grep LD_ASSUME_KERNEL and it returns nothing..:( And just for confirmation : I had the same problem as Lee had (unable to make calls out) :( Regards, Stevanus Lee Archer wrote: Yes it is a problem cos after a while of just leaving it the system is unable to make calls out via the PSTN, which is why I have spent time with the telco, more like wasted time, and played with zaptel's make options. After trying a few things I came to the temporary conclusion that it was the zaptel watchdog trying and failing to restart a hung channel. I recompiled zaptel without the watchdog and a few days later it did the same so I'm back to sq 1. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dave Cotton Sent: 18 April 2006 09:27 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] multiple asterisk process ? On Tue, 2006-04-18 at 09:13 +0100, Lee Archer wrote: Any thoughts as to why only 1 of my boxes has this problem? Is it really a problem? I'm on a 2.6 kernel so any more ideas? Can someone answer what was the original purpose of the "export LD_ASSUME_KERNEL=2.4.1" in the asterisk script? Perhaps Gregory Boehnlein, the author, will be able to tell us. -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] multiple asterisk process ?
Hmm...my output for getconf GNU_LIBPTHREAD_VERSION is NPTL 2.3.4.. I don't know what it's mean anyway :P And for Lee, I'm configuring my asterisk through amp (now freepbx), but I do some custom configuration manually too ;) I guess Paul is right, I suspect there are bugs in asterisk that haven't been solved like "avoiding deadlock on iax" problem which I had mentioned before.. Unfortunately, I don't know how to recreate the problem so all I can do if the problem is happened just do some killall - 9 asterisk :(... Regads, Stevanus Moises Silva wrote: Thanks for clarifying that Paul. my output for getconf is: linuxthreads-0.10 so i guess is "normal" to have several threads shown by "ps axu" right? On 4/17/06, Dave Cotton <[EMAIL PROTECTED]> wrote: On Mon, 2006-04-17 at 19:12 +0200, Paul Hewlett wrote: This is incorrect. Asterisk is a multithreaded system but how the threads are handled by the OS depends on the version of threads that is being used. For Linuxthreads (kernel 2.4), one would see a separate entry for each thread when executing 'ps aux'. For NPTL (linux 2.6) one does not see each thread as a separate entry. So the OP must tell us which kernel version he is using. Alternatively type getconf GNU_LIBPTHREAD_VERSION as root. For NPTL u should get something like NPTL 2.3.5 or suchlike. If you are using NPTL and there is more than one entry for asterisk, then asterisk has spawned an extra process for some reason. If extra processes keep appearing then I would say that he has a bug or error somewhere and asterisk is respawning that separate process. Are you sure? root 2532 0.0 0.2 2532 620 ?S17:22 0:00 /bin/sh /usr/sbin/safe_asterisk root 2539 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2542 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2544 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2545 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2546 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2547 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2548 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2549 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2550 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2551 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2552 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2553 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2554 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2555 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2556 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2557 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2558 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2559 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2560 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2561 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2562 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2563 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2564 0.0 2.8 17716 7316 ?S17:22 0:01 asterisk -n -vvvg -c root 2565 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2566 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c root 2567 0.0 2.8 17716 7316 ?S17:22 0:00 asterisk -n -vvvg -c With NPTL 2.3.6 If that is the case * is totally hosed, no? -- Dave Cotton <[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] multiple asterisk process ?
Hi, Why does my asterisk keep forking instances at random times everyday? When I do ps aux, I got this: asterisk 13068 2.2 5.1 25924 12276 ? Sl 06:00 13:18 asterisk -vvvg -c asterisk 23558 0.0 5.1 26040 12248 ? S09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S12:32 0:00 asterisk -vvvg -c asterisk 2520 0.0 5.1 25928 12228 ? S13:21 0:00 asterisk -vvvg -c asterisk 4638 0.0 5.1 25924 12232 ? S13:50 0:00 asterisk -vvvg -c asterisk 5126 0.0 5.1 25932 12240 ? S13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Problem
It seems that your asterisk cannot transcoding from ulaw to g729 and vice versa. What is the output from 'show translation' ? Did you allow both codecs in sip.conf ? Regards, Stevanus Il Neofita wrote: I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this error Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoiding initial deadlock on iax?
Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 WARNING[8386] channel.c: Avoided initial deadlock for '0x81d9530', 10 retries! It happens unpredictably and all I can do just killall -9 asterisk :S. When I execute iax2 show channels on CLI, I got messages that indicate many iax channel hung and I cannot do soft hangup to them :(. Here is my iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) tos=0x68 ; bandwidth=low jitterbuffer=yes dropcount=2 disallow=all allow=ilbc ;allow=g723.1 ;allow=g729 ;allow=ulaw ;allow=alaw ;allow=gsm mailboxdetail=yes the other settings on iax.conf are just iax2 account for trunk and personal use. So I cut them in order to save spaces... Perhaps it's a bug? I've found this http://bugs.digium.com/view.php?id=4045 , but from the link I read that it is just for H323 not for iax. Will that patch cure my asterisk problem since the symptom are the same? Anyone has any ideas? Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk hang
Thanks for the reply.. Here is the output from ps aux when asterisk is hung: asterisk 7219 2.5 4.2 22144 10104 ? Sl Mar15 72:36 asterisk -vvvg -c asterisk 29444 0.0 4.5 25180 10820 ? S Mar16 0:00 asterisk -vvvg -c asterisk 29445 0.0 4.5 25180 10820 ? S Mar16 0:00 asterisk -vvvg -c asterisk 29460 0.0 4.5 25184 10880 ? S Mar16 0:00 asterisk -vvvg -c asterisk 30681 0.0 4.6 25268 11208 ? S Mar16 0:00 asterisk -vvvg -c asterisk 31378 0.0 4.7 25332 11380 ? S Mar16 0:00 asterisk -vvvg -c asterisk 32665 0.0 4.8 25264 11492 ? S Mar16 0:00 asterisk -vvvg -c asterisk 1187 0.0 4.9 25484 11888 ? S Mar16 0:00 asterisk -vvvg -c asterisk 3218 0.0 5.0 25540 12048 ? S Mar16 0:00 asterisk -vvvg -c asterisk 13014 0.0 5.0 25544 12084 ? S Mar16 0:00 asterisk -vvvg -c asterisk 25953 0.0 5.0 25536 12124 ? S Mar16 0:00 asterisk -vvvg -c asterisk 25954 0.0 5.0 25536 12124 ? S Mar16 0:00 asterisk -vvvg -c asterisk 25955 0.0 5.0 25536 12124 ? S Mar16 0:00 asterisk -vvvg -c I didn't run two asterisk. You can see from the output above that I ran one processes of asterisk on 15th March. The other processes just came out by themselves. I really don't know where they are from..:S I didn't get any messages when asterisk is hung :(. I could even connect to the CLI using asterisk -r as if asterisk worked properly. But when asterisk is in this state, sometimes I cannot make outgoing call (not even sip-to-sip call) and sometimes I lose the dial tone at all.. Pretty weird.. Regards, Stevanus Tzafrir Cohen wrote: On Wed, Mar 15, 2006 at 11:01:21AM +0700, stevanus wrote: Hi, Does anyone experience asterisk hang with awfully a lot of processes (asterisk -vvvg -c)? Do you run two of them? What are the messsages that you see when it is "hung"? Any chance you sent one to the background using ctrl-Z? My asterisk is often hang at unprecedented times and the only thing I can do is killall-9 asterisk.. After I do that, asterisk is come back to normal again... What is the cause for this? I use asterisk 1.2.5, zaptel 1.2.4 and 3 TDM04B on my machine.. Is it possibly caused by interrupt sharing? I admit I get some interrupt sharing problems and cannot be solved right now because of lack of funds :(.. Interrupts sharing may cause problems with the quality of sound. But will not get asterisk stuck. Please provide more data. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk hang
Hi, Does anyone experience asterisk hang with awfully a lot of processes (asterisk -vvvg -c)? My asterisk is often hang at unprecedented times and the only thing I can do is killall-9 asterisk.. After I do that, asterisk is come back to normal again... What is the cause for this? I use asterisk 1.2.5, zaptel 1.2.4 and 3 TDM04B on my machine.. Is it possibly caused by interrupt sharing? I admit I get some interrupt sharing problems and cannot be solved right now because of lack of funds :(.. Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] intel 536 ep as fxo -> possible?
Hi, Sorry for keep hammering the list with this annoying question. Can we use Intel 536 ep (not 537ep that is in wiki) as x100p clone? I know I've asked it in this list a couple days ago but none responded so far and I'm getting frustrated pairing it with asterisk as the zaptel driver could not detect it. I just need more information before I throw this intel 536 EP to the garbage can :P. Any information would be appreciated.. Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] intel 536 EP as x100p clone?
Hi.. I have one intel 536 EP. Does it possible use it as x100p clone for asterisk? I tried today with no luck :(.. Here is what I did : - plugged the card the card is recognised as (lspci -vv): 00:09.0 Communication controller: Intel Corp. 536EP Data Fax Modem Subsystem: Intel Corp.: Unknown device 1000 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- SERR- Latency: 32, Cache Line Size 08 Interrupt: pin A routed to IRQ 12 Region 0: Memory at e200 (32-bit, non-prefetchable) [size=4M] Capabilities: [e0] Power Management version 2 Flags: PMEClk- DSI- D1- D2+ AuxCurrent=375mA PME(D0-,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable+ DSel=0 DScale=0 PME- - make and make install zaptel driver - modprobe zaptel (there's no output) - modprobe wcfxo and get these following messages: ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxo Is there any possibilities using this card with asterisk? I've searched wiki and there I found that people are success using Intel 537EP as x100p clone. While mine is merely Intel 536EP, but I think both are modems made by Intel.. Maybe there's a way making it function like x100p too like someone in asterisk channel (irc) told me :).. Thanks.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.3 call problem
Hi, Had checked the log and found this : Jan 26 12:18:19 WARNING[15491] chan_iax2.c: Unable to create translator path for unknown to g723 on IAX2/trunk-4 Hmm that's strange cause in my iax.conf, I allowed only ilbc in both ends. Is this a bug in asterisk 1.2.3 or a feature? Seems like codec negotiation algorithm problem... Here is my iax.conf for both ends : [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) tos=0x1c ;bandwidth=low ;jitterbuffer=yes ;dropcount=2 disallow=all allow=ilbc ;allow=g723.1 ;allow=g729 ;allow=ulaw ;allow=alaw ;allow=gsm mailboxdetail=yes #include iax_additional.conf #include iax_custom.conf Regards, Stevanus stevanus wrote: Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ? Please shed me some light, thank you.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk 1.2.3 call problem
Hi, I've tried to upgrade my asterisk to 1.2.3 again after disastrous bug incident yesterday but when I called and the phone was picked up, there was simply busy tone... Weird, is this another bug in asterisk 1.2.3? Currently, I rollback again to asterisk 1.0.10...:( Is there any configuration change issue in 1.2.3 cause I've just used my configuration that worked in asterisk1.2.2 ? Please shed me some light, thank you.. Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] jitterbuffer causes no sound?
Hi guys, I 've tried asterisk 1.2.2. It work flawlessly for about 3 days then at the third days I activated setting jitterbuffer=yes and suddenly there is no voice when the call is picked up. It's really weird as if asterisk stops sending rtp packet. I've checked asterisk log and found nothing suspicious. Just weird :S. I tried it in 3 asterisk server and all of them are having the same symptoms (i.e: no voice). There is no sound when the call is pickup, no matter the call is from sip to sip, sip to zap, zap to sip ,sip to zap through iax, nor sip to sip through iax... Is jitterbuffer really the culprit or it's just a coincidence that I activated the jitterbuffer and my asterisks stopped working? Is asterisk 1.2.2 not meant for production use? Has there someone success story implemented asterisk 1.2.2? If there's, please share me as it can encouraged me to try this beast again :)... Currently, I'm rollback to asterisk 1.0.10 to avoid any unprecedented issue... Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] weird zttest result
Is this result indicates no problem at all? 8192 samples in 27554 sample intervals -136.352539% Regards, Stevanus C F wrote: These are actulay not strange, but good results. On 1/23/06, stevanus <[EMAIL PROTECTED]> wrote: Hi, I have these strange results : 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 27554 sample intervals -136.352539% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% Anyone has any idea why this happens? Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] weird zttest result
Hi, I have these strange results : 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 27554 sample intervals -136.352539% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% Anyone has any idea why this happens? Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speex in asterisk 1.0.10
Yeah, Paul. I guess you're right.. Just tested speex and got complains from my customer :S..Maybe this codec is not suited for our network ;).. Regards, Stevanus [EMAIL PROTECTED] wrote: Quick question - what is the point of speex? Do we really need it as an option? PaulH - Original Message - From: "stevanus" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Thursday, January 19, 2006 3:37 PM Subject: [Asterisk-Users] speex in asterisk 1.0.10 Hi, Does anyone know how to configure speex in asterisk 1.0.10? I've successfully installed it but cannot get any idea how to set the quality, etc.. Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] speex in asterisk 1.0.10
Hi, Does anyone know how to configure speex in asterisk 1.0.10? I've successfully installed it but cannot get any idea how to set the quality, etc.. Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2.2 Released!
Just curious, will this version be supported by AMP 1.10.010? Anyway, I am going to upgrade mine in saturday..:) Joe Pukepail wrote: Perhaps I'm an idiot, but I looked through the readme and changelog but can't figure out what asterisk-netsec is all about? Anybody figure it out? On 1/18/06, Mr. James W. Laferriere <[EMAIL PROTECTED]> wrote: Hello Announce & All , On Wed, 18 Jan 2006, Asterisk Development Team wrote: > Greetings everyone! > The 1.2.2 versions of Asterisk, Zaptel, and Libpri have now been > released. The source tarballs are available for download on > ftp.digium.com. For details about what has changed, see the ChangeLog > for Asterisk, Zaptel, or Libpri. > > We are also excited to announce the release of a special version of > Asterisk 1.2.2, called Asterisk-NetSec. It includes some very exciting > features not available in any other version of Asterisk, or even any > other related product! Please view the appropriate README and ChangeLog > for more details. > > Asterisk-addons and Asterisk-sounds will remain at version 1.2.1. > Previously, all packages were updated to reflect a matching version > number, even if no changes have been made. From now on, releases will > only be made when changes have actually been made. Even if version > numbers do not match, it is safe to use all of these releases together, > as long as all of them are the latest version available. > > Thank you! The one thing that annoys me most is a announcment with out a url: to what it is announcing . Can we please correct this ? Tia , JimL ps: Not that I can't find it , but ... is just courtisy to others . -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | | http://www.asteriskhelpdesk.com/cgi-bin/astlance/r.cgi?babydr | +--+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0.10 to 1.2.1 upgrade..is it worth it?
Hi, As I've dealt with asterisk 1.0.10 successfully, I wonder what the benefit I will get from upgrading to 1.2.1.. [Of course I know there're lot of new interesting stuffs in 1.2.1, but are they stable already?] Does the 1.2.1 need more resources, more power hungry? Anyone has success story with asterisk 1.2.1 please share :) Thank you... Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ip phone
Hi, Maybe grandstream budgetone 100 series will fulfill your requirement. It's very good for such a cheap sub-50 phone. Once, I've tested and I've found myself that it's a good performer (even it has compatibility problem with old switch in my office :P) You can search the supplier through googling it. Don't ask me as I don't know any information about it. Good luck.. Regards, Stevanus trixter aka Bret McDanel wrote: looking for ip phones for an office setting. The client wants about 15 phones initially. Not counting volume discounts, does anyone have any recommendations. Cost is a factor, after discounts they were thinking about $50/phone. The following came up that seem to fit, any experiences with these models would be requested, any that arent on this list would alsso be recommended providing they fit somewhere around the price guideline. most of what is on http://www.voipsupply.com/index.php?cPath=95_105 qualifies for what I am looking for, I just wanted something other than someone who stands to profit off the sale to give personal experiences :) Looking for very good audio quality, no discernable echo, etc. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing busy
Hi, Yeah, you're right Olle. The connection to another sip provider is set in sip.conf. I thought Anders was trying to make outgoing call to pstn. Too much tinkering with Digium card make me think that :P. Regards, Stevanus Olle E. Johansson wrote: stevanus wrote: Hi, Outgoing setting is in zapata.conf. I think you should read the wiki more ;). If what you mean by outgoing is another sip extension then you should look for extension.conf. Links: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf And in fact it was all about sip.conf ;-) As you say, connecting to SIP service providers is well documented on the wiki, but not on those pages. /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outgoing busy
Hi, Outgoing setting is in zapata.conf. I think you should read the wiki more ;). If what you mean by outgoing is another sip extension then you should look for extension.conf. Links: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf http://www.voip-info.org/tiki-index.php?page=Asterisk+config+extensions.conf Regards, Stevanus Anders Svensson wrote: I have a problem. Incoming calls work without problem but I cant call out. Using AAH.Gets a busy tone Anyone who can see a mistake in Outgoing settings context=from-pstn host=ipkund1.rixtelecom.se insecure=very nat=yes secret=xxx type=peer username=0406082250 Regards Anders Svensson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk frequently dead
Hi, Thanks for the reply.. But the biggest problem here is that people using asterisk will get dissapointed, sometimes mad because their call being dropped off when asterisk is dead.. Any suggestions anyone? Regards, Stevanus Moises Silva wrote: this is not a solution, more a workaround, you can try using svscan service, so when down will automagically briged up. On 9/15/05, stevanus < [EMAIL PROTECTED]> wrote: Hi, I've tried upgrade my asterisk to 1.0.9... It's now seemed that asterisk is more stable but it's still dead by itself occasionally.. Output from gdb yield this: ... Reading symbols from /lib/libgcc_s.so.1...done. Loaded symbols for /lib/libgcc_s.so.1 #0 0x00a597a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 (gdb) ... Actually the information giving by gdb is far more detail..Just keep it brief here to keep the space small. If anyone want to help me, then I'll send it entirely.. Any comments/thoughts will be greatly appreciated. Thanks, Best regards, Stevanus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org" ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk frequently dead
Hi, I've tried upgrade my asterisk to 1.0.9... It's now seemed that asterisk is more stable but it's still dead by itself occasionally.. Output from gdb yield this: ... Reading symbols from /lib/libgcc_s.so.1...done. Loaded symbols for /lib/libgcc_s.so.1 #0 0x00a597a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 (gdb) ... Actually the information giving by gdb is far more detail..Just keep it brief here to keep the space small. If anyone want to help me, then I'll send it entirely.. Any comments/thoughts will be greatly appreciated. Thanks, Best regards, Stevanus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk frequently dead
Hi,
Sorry about the lack of information...
I use RHEL 4, asterisk cvs stable v1.0 and compile it myself..
It was worked well..
Asterisk was run stable in old platform (use duron), but then when I
upgraded it to P4, the problem is exists.
The weird things is I set asterisk in the same exact machine and the
problem only lies in this one..
The others run stable.
Maybe it's because I do share interrupt for asterisk? Will it help for
asterisk stability?
Here is output from lspci:
00:00.0 Host bridge: Silicon Integrated Systems [SiS]
661FX/M661FX/M661MX Host (rev 11)
00:01.0 PCI bridge: Silicon Integrated Systems [SiS]: Unknown device
0003
00:02.0 ISA bridge: Silicon Integrated Systems [SiS] SiS964 [MuTIOL
Media IO] (rev 36)
00:02.5 IDE interface: Silicon Integrated Systems [SiS] 5513 [IDE] (rev
01)
00:02.7 Multimedia audio controller: Silicon Integrated Systems [SiS]
Sound Controller (rev a0)
00:03.0 USB Controller: Silicon Integrated Systems [SiS] USB 1.0
Controller (rev 0f)
00:03.1 USB Controller: Silicon Integrated Systems [SiS] USB 1.0
Controller (rev 0f)
00:03.2 USB Controller: Silicon Integrated Systems [SiS] USB 1.0
Controller (rev 0f)
00:03.3 USB Controller: Silicon Integrated Systems [SiS] USB 2.0
Controller
00:04.0 Ethernet controller: Silicon Integrated Systems [SiS] SiS900
PCI Fast Ethernet (rev 90)
00:08.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
00:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
00:0a.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface
01:00.0 VGA compatible controller: Silicon Integrated Systems [SiS]
661FX/M661FX/M661MX/741/M741/760/M760 PCI/AGP
And here is output from cat /proc/interrupts
CPU0
0: 666453883 XT-PIC timer
1: 16 XT-PIC i8042
2: 0 XT-PIC cascade
5: 666283487 XT-PIC ohci_hcd, wctdm
8: 1 XT-PIC rtc
9: 0 XT-PIC acpi, ehci_hcd
10: 666272466 XT-PIC SiS SI7012, ohci_hcd, wctdm
11: 697289929 XT-PIC ohci_hcd, wctdm, eth0
12: 66 XT-PIC i8042
14: 1557961 XT-PIC ide0
15: 1557358 XT-PIC ide1
NMI: 0
ERR: 0
Already do make clean, make and make install in the new platform.
Seems do not help at all...
* sigh *
Can you pinpoint what causes it to crash?
This is a tough question...I have no idea of what causing this or what should I do right now...
Perhaps somebody willing to give me 5 minutes tutor of using gdb?
I'm in process of learning it...Gotta be careful cause the system is used by more than 10 person (well, I'm getting tired of apologizing anyway :P)
Thanks,
Best Regards,
Stevanus
Andrew Kohlsmith wrote:
On Wednesday 07 September 2005 22:56, stevanus wrote:
My asterisk is frequently dead by itself.
It leaves messages:
/usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Someone new who's left us a wealth of information so we can diagnose the
problem quickly and help him find a timely solution.
When you take your car to the mechanic, do you simply say "It's broken. It
doesn't run the way it should." or do you give him some details. In this
case:
- Distribution of Linux
- Source of your Asterisk binaries (distribution packages, did you compile
yourself?)
- Version of Asterisk
- Has it ever worked
- Can you pinpoint what causes it to crash
I mean honestly, how do you expect us to help?
-A.
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[Asterisk-Users] asterisk frequently dead
Hi,
My asterisk is frequently dead by itself.
It leaves messages:
/usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
Anyone has any idea of the cause?
Thanks..
Best Regards,
Stevanus
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Re: [Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Hi,
I have similar problems like you.
In the past, I did adjusted my RX and TX gain, but didn't know if it has
been optimal yet.
Fxotune is seemed do not working, perhaps caused of my asterisk's
version ( I use stable v1.0)..
Just curious, is rx and tx gain really a sole setting option here in
order to make things the way it's meant to be? Or is there others?
FYI, my tdm04b occasionally don't detect call-in as well as hangup signal.
I've searched in the wiki and have activated hanguponpolarity swicth.
But I don't notice any difference at all.
Any help would be greatly appreciated. (I've asked this in another
thread, but got no respon :( )
Best Regards,
Stevanus
canuck15 wrote:
This may or may not be related but have you tried adjusting your RX and TX
gains? I see both are at the default (0.0) which leads me to believe you
have not. Search the Asterisk Wiki for the procedure.
-Original Message-
From: Faris Raouf [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 07, 2005 12:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] TDM400P not detecting hangup and not hanging up.
Can anyone suggest where I might begin looking for an answer please?
I have just installed a TDM400P (2x FXS and 1x FXO modules installed)
The first problem is that it does not seem to be able to detect if the
remote party has hung up when a call comes through on the FXO. For example,
if someone calls in, and then hangs up at any time after it starts ringing,
Asterisk carries on as though the caller never hung up.
I've tried raising BATT_THRESH to 8 in wcfxs.c and re-compiling zaptel (this
was the only thing that Google came up with to help me, although others do
seems to have had similar problems to mine at various times), but it has
made no difference at all.
The second problem is that Hangup does not hangup. The channel stays open
until I stop asterisk.
Note: When MAKING a call on the FXO, when I terminate the call on my SIP
phone the line does drop correctly. The problem appears to be related to
incoming calls only.
I'm based in the UK. Using RedHat 9 with zaptel-1.0.9.1, asterisk-1.0.9 (and
chan_capi-0.5.4)
Thanks in advance for any ideas.
Faris.
*
Here's my initialisation script:
modprobe zaptel
modprobe wctdm opermode=UK
/sbin/ztcfg -
capiinit
safe_asterisk
zapata.conf
[trunkgroups]
; nothing in here
[channels]
rxwink=300 ; (I tried commenting this out. Make no difference)
usedistinctiveringdetection=no
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
usecallingpres=no
sendcalleridafter=1
callwaitingcallerid=no
threewaycalling=no
transfer=no
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=no
rxgain=0.0
txgain=0.0
immediate=no
progzone=uk
; module 0 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 1
; module 1 on card is an FXS
signalling=fxo_ks
language=en
context=sip
channel => 2
; module 2 on card is an FXO
signalling=fxs_ks
language=en
context=faris
channel => 3
zaptel.conf
fxoks=1-2
fxsks=3
loadzone=uk
defaultzone=uk
and in extensions.conf
[faris]
exten => s,1,NoOp(cid=${CALLERID})
exten => s,2,Wait(10)
exten => s,3,Answer
exten => s,4,Wait(1)
exten => s,5,Playback(some-long-message) exten => s,6,Hangup
The long wait(10) is just there to see what happens. Removing it makes no
difference. Basically whenever a call comes in, no matter when the caller
hangs up, Asterisk continues with the call to the end (i.e. plays long
message).
What's more, the Hangup at the end has no effect. The line is not dropped.
The line is not ever dropped in fact.
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Re: [Asterisk-Users] tdm04b hangup problem
Hi, I'm sorry about the false information. It seems after the crash, the problems is still exist. Anyone can help me? Could it be IRQ issue? Here is output from cat /proc/interrupt: CPU0 0: 92807252 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 5: 92732654 XT-PIC ohci_hcd, wctdm 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi, ehci_hcd 10: 92730937 XT-PIC SiS SI7012, ohci_hcd, wctdm 11: 95761662 XT-PIC eth0, ohci_hcd, wctdm 12: 66 XT-PIC i8042 14: 163276 XT-PIC ide0 15: 997605 XT-PIC ide1 NMI: 0 ERR: 0 Best Regards, Stevanus stevanus wrote: Hi, Yesterday, the asterisk machine was crash :S. But after the crash, it seems the previous problems were eliminated. I will notice it in about a week or two. If it's stable now, so the recommended solution when there are problems with asterisk is to restart the machine? Weird. Does nobody like to share any comments? Just curious :P Best Regards, Stevanus stevanus wrote: Any thought anyone? stevanus wrote: Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming call so I have to watch fop all the time and hangup the channel manually there. Another problem is when an outgoing call was placed and the caller ended the conversation, the tdm04b did not hangup the channel. So when the caller does off hook too fast and interpreted by asterisk as hold, both zap channel will be connected by asterisk as the caller hangup the second call. Anyone experiences this issue? Is it possible that this is caused by improper setting in rxgain or txgain? Currently, I set rxgain = 15.0 and txgain = 5.0.. Thanks.. Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm04b hangup problem
Hi, Yesterday, the asterisk machine was crash :S. But after the crash, it seems the previous problems were eliminated. I will notice it in about a week or two. If it's stable now, so the recommended solution when there are problems with asterisk is to restart the machine? Weird. Does nobody like to share any comments? Just curious :P Best Regards, Stevanus stevanus wrote: Any thought anyone? stevanus wrote: Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming call so I have to watch fop all the time and hangup the channel manually there. Another problem is when an outgoing call was placed and the caller ended the conversation, the tdm04b did not hangup the channel. So when the caller does off hook too fast and interpreted by asterisk as hold, both zap channel will be connected by asterisk as the caller hangup the second call. Anyone experiences this issue? Is it possible that this is caused by improper setting in rxgain or txgain? Currently, I set rxgain = 15.0 and txgain = 5.0.. Thanks.. Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] tdm04b hangup problem
Any thought anyone? stevanus wrote: Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming call so I have to watch fop all the time and hangup the channel manually there. Another problem is when an outgoing call was placed and the caller ended the conversation, the tdm04b did not hangup the channel. So when the caller does off hook too fast and interpreted by asterisk as hold, both zap channel will be connected by asterisk as the caller hangup the second call. Anyone experiences this issue? Is it possible that this is caused by improper setting in rxgain or txgain? Currently, I set rxgain = 15.0 and txgain = 5.0.. Thanks.. Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tdm04b hangup problem
Hi, I have severe problem here.. My asterisk server use tdm04b from digium and is often incapable of detecting hangup signal. It is happened occasionally in incoming call so I have to watch fop all the time and hangup the channel manually there. Another problem is when an outgoing call was placed and the caller ended the conversation, the tdm04b did not hangup the channel. So when the caller does off hook too fast and interpreted by asterisk as hold, both zap channel will be connected by asterisk as the caller hangup the second call. Anyone experiences this issue? Is it possible that this is caused by improper setting in rxgain or txgain? Currently, I set rxgain = 15.0 and txgain = 5.0.. Thanks.. Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sccp help
Hi, Haven't noticed that there exists one :P Thanks for the pointer anyway ;). Gotta sign up pretty soon :) Best Regards, Stevanus Stefan Gofferje wrote: On 10:10:54 August 19, 2005 stevanus <[EMAIL PROTECTED]> wrote: Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Jep, it is... If you had problems with this, your chance for a solution is higher at the chan-sccp-users list... :-) Regards, Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sccp help
Hi, I used chan_sccp from ftp://ftp.berlios.de/pub/chan-sccp. Is it the same as chan_sccp from chan-sccp.berlios.de? Best Regards, Stevanus Stefan Gofferje wrote: Hi, On 9:04:57 August 19, 2005 stevanus <[EMAIL PROTECTED]> wrote: Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : I've tried different versions of chan_sccp, yet the result were still the same. Which version of chan_sccp did you use? Sourceforge or Berlios? There is a new fork of chan_sccp by Sergio Chersovani who started work some weeks ago and did an almost complete rewrite of the channel. This version supports a lot more features on various phones and has a lot less bugs. You could find it at chan-sccp.berlios.de (official site) or chan-sccp.org (unofficial site). There is a related mailinglist at berlios.de where Sergio does a hell of a lot of support (unless he is one vacation like at the moment :-) ) and gladly accepts bug reports :-). Regards, Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sccp help
Hi, I tried to connect cisco 7910 into asterisk system using chan_sccp.so. But I got a major issue : - when I called from 7910 to another sip phone in the same asterisk server, the call took place normally. - when I called from 7910 to another sip phone in different asterisk server, the call is answered but I cannot hear nor say anything. The phone just immediately lose its tone. - when I got a call from another sip phone in the same asterisk server, the phone rang. But after I picked the handset, there were no tone at all.. sccp debug on CLI produced the following messages: SCCP: Alarm Message: Severity: Major (7), 29: DSP Keepalive Timeout [0x5, 0xa, 0x8, 0x2](5) [21/1090360010] I've tried different versions of chan_sccp, yet the result were still the same. Is it time for me to dump this cisco phone to the garbage can ? (I hope not) Anybody had experienced similar issues? Any suggestion will be greatly appreciated.. Thanks Best Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] delay problem
Hi, I've experienced excessive delay when called from one extension number to another... This happened unstable, as the delay range between 2 - 20 seconds... I'm using Duron 950 MHz with memory 256 MB as asterisk server and my asterisk currently serves 30-40 accounts.. Concurrent calls vary between 1-10 calls. Is my Duron overwhelmed by the load? The delay exists in queue, local sip-to-sip call, and zap-to-sip call. It's so annoying :( Anyone has a solution or maybe some clue for me? Totally clueless here... Thanks... Regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No sound
Hi, That would probably be a problem with nat. Just read this on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+SIP+NAT+solutions Best regards, Stevanus Ronald Wiplinger wrote: I have an asterisk box installed, but all connections to outside of the private network do not have a sound. Can you give me a hint what it could it be? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco 7940 + sccp issue
Stefan Gofferje wrote: "If the phone just requests CTLSEPxxx.tlv and nothing else, it either have been used on a CallManager with authentication / encryption enabled and is now security locked because the asterisk does not provide the proper tlv-file or the firmware is corrupted. Try to reset to factory settings. if this does not help, try to reflash the firmware. " Hi, I've unlocked the phone by pressing **# and set it back to factory setting. But the problem still exists. Do I really need to reflash the phone? Sorry just wanna assure myself that the action is necessary in order to make my 7940 talk with asterisk using sccp. I had bad experiences in flashing devices therefore I want to avoid this as much as possible :). Best regards, Stevanus Stefan Gofferje wrote: Hi, On 9:20:51 July 06, 2005 stevanus <[EMAIL PROTECTED]> wrote: I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP.tlv from my tftp server. If this does not help - well shit happens... Just kidding... :-). If you have a legal license for the phone software, you could send the phone to Cisco if nothing else helps. Regards, Stefan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco 7940 + sccp issue
Hi, Does anyone know how to make this thing (7940) work with asterisk (chan_sccp module) ? I've set the configuration according to the wiki and now the phone just keep asking for CTLSEP.tlv from my tftp server. In the cisco's web interface, I found this in the device logs : 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 0x8106, 0x0, 0x12300800 ... I've already tried all the trick from wiki, still yield no result :(.. Any help would be greatly appreciated. Thanks... Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zeroconf help
hi, recently I installed zeroconf for asterisk... I've already followed the asterisk+zeroconf how to (which is too short), but it came with an error message... asterisk: relocation error: /usr/lib/asterisk/modules/res_zeroconf.so: undefined symbol: DNSServiceRegister Ouch ... error while writing audio data: : Broken pipe it's weird since I've double checked the library and header from zeroconf and it seems that everything has been in the right place.. Is there anyone can help me? Well, it seems I hit another dead end this time... Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to allow multiple codecs in A@H
hi, just put those lines (allow=bla) in peer details box in AMP GUI. at section "add sip trunk". best regards, stevanus Erdem HAKİ wrote: I wonder how to allow more then one codec in AMP ([EMAIL PROTECTED]) GUI? For example I want to configure like this allow=gsm allow=g729 ... I can add these by editing sip_additional.conf, but i want to add codecs using AMP, any suggestions? Thanks Erdem HAKI – [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729AB codec support
Hi, As far as I remember, Intel sources state that noone has to spend anything on their kind of implementation of G729, if he or she doesn't get any benefit from it at all (i.e: non-commercial use). Personally, I agree for this kind of practice as this will speed education rate for people who cannot afford it. [In my country, there are still many IT professionals that is paid under $200 per month :( ] Best regards, Stevanus Zoa wrote: Im saying that the code is only an implementation of g729. The intel sources clearly states that you need a license for g729, not from intel but from the g729 patent holder. Zoa. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone for Linux desktops
Hi, try xlite, it has linux version.. Best regards, Stevanus Eric Bishop wrote: Hi all, We are successfuly running an Asterisk server with standard SIP hard phones and it is working well. We are looking to deploy some soft phones on our Linux desktops. There seems to be several floating about. Anyone out there with some good/bad experiences with particular Linux softphones. We only need g711 and prefer IAX but a SIP one will do Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bypass incoming ring..is it possible?
Hi, I've tried your suggestion but the result is still the same... Have another suggestion? Best regards, Stevanus Wilson Pickett wrote: Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did? Try setting callerid=no on the FXO channel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Thank you for the timely suggestion
Hi, try xlite if you have enough bandwitdh for G711 codec requirement.. try firefly if you want to use G729 codec freely (linked via dll).. both of them are the best freeware softphone for windows. Best regards, Stevanus infra struct wrote: I have been searching for the necessary components for my setup from sometime back; yet to install Asterisk and will be installing softphones on Linux Server and on all windows PCs(most of them are Windows Xp,others are Windows 2000 professional,Windows 98); but could not decide which softphone to use still searching for the softphone.. Discover Yahoo! Stay in touch with email, IM, photo sharing & more. Check it out! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] tdm04b slow response
Hi, After days tinkering with this digium card (TDM04B), I notice that this card has a slow response in detecting ring signal from pstn and hanging up when the call is over. The delay can consume up to several seconds... Is this normal? Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ivr not working?
Finally I've got the ivr to work.. The workaround I've found so far is record and edit the voice file through Adobe Audition or cool edit or sound recorder, etc and then convert it to gsm using sox.. Hope that might help someone ;) Best regards, Stevanus stevanus wrote: Hi, Recently, I've just installed asterisk along with AMP.. Everything seems to work fine, just when I tried to record my voice via ivr, asterisk won't play the file if I call it. When I test by dialing *99, the record is played, but when I call straight to the digital receptionist, it just stand there about 7 seconds, playing no sound at all and then hung up.. I use AMP version 1.10.007a.. Has anyone known the solution for my problem? Any help would be appreciated a lot.. Thanks.... Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] bypass incoming ring..is it possible?
Hi, I've tried your suggestion but it still yield no result :( Personally, I think the problem lies in zaptel driver, because when I see the interactive asterisk log (as I set asterisk to run in verbose mode) , any call to the asterisk box will result in one ring then the digium TDM04B start to react by giving line such as - Starting simple switch on Zap1/1 -. Then there will be a moment of pause. By the time the asterisk answering the call, that would be another ring. :( [I've already put line : exten => s,1,Answer at pstn context] I've looked at /etc/zaptel.conf and /etc/asterisk/zapata.conf but couldn't find any setting I needed so far :(.. Still found a dead end... Maybe I must modify the zaptel driver that comes from the CVS, am I right? Well, yesterday I tried to read some of the sources and they were so overwhelming [I'm not an expert C programmer].. I believe it will take some weeks, perhaps months to learn the source codes Perhaps I should take a C course as well :P Anyone can help me? Best regards, Stevanus Alexander Ilyushin wrote: You can first answer to call, and then provide playtones(ring) to caller. 2005/6/8, stevanus <[EMAIL PROTECTED]>: Hi, Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did? Is anyone able to give me any clues or pinpoint me where I can get more information about it? Thanks for your attention.. Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bypass incoming ring..is it possible?
Hi, Is it possible to bypass incoming ring on asterisk so that incoming calls come to asterisk box will be directed straight into did? Is anyone able to give me any clues or pinpoint me where I can get more information about it? Thanks for your attention.. Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Phone always busy after caller hangup
Hi, If you use digium card, then maybe you set wrong signaling on fxs... Best regards, Stevanus Tim P wrote: I have multiple sipura 2100 boxes connected to my * box and for some reason that i cannot figure out when making a call to one and answering it and then hanging up results in the line be permanently busy (the phone called is permanently busy until * is rebooted). Any idea where to start with this one? It seems to me that either the SPA2100 is not registering the end of the call or * isn't. I suspect the SPA2100 but see nothing in the logs or in the SPA config to indicate a fix. Any ideas? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ivr not working?
Hi, Recently, I've just installed asterisk along with AMP.. Everything seems to work fine, just when I tried to record my voice via ivr, asterisk won't play the file if I call it. When I test by dialing *99, the record is played, but when I call straight to the digital receptionist, it just stand there about 7 seconds, playing no sound at all and then hung up.. I use AMP version 1.10.007a.. Has anyone known the solution for my problem? Any help would be appreciated a lot.. Thanks Best regards, Stevanus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
