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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

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

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

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<xxx>.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 ...

Dear all. I have install samba 2.2.7a in mandrake 9.1 and success access by client, but there is a problem about it. Server samba is giving an ip address 192.11.11.xxx and client is having ip address 192.11.11.xxx and 192.11.12.xxx. in computer client (windows NT, Win2k, and win98) which have an ip address 192.11.11.xxx server can access straightly from network neighborhood (without searching

Dear all, Here is my code which am using to combine 5th column from different data sets. Here is the function to do my job genesymbol.append.file <-NULL gene.column <- NULL readGeneSymbol <- function(files,genesymbol.column=5){ for(i in fnames){ temp <- read.table(fnames,header=T,sep="\t",stringsAsFactors=F,quote="\"")

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

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

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

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

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} </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

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 ?

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-

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

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 <ip address>:16480: Network is unreachable May 2 09:38:29 WARNING[17120] chan_sip.c: sip_xmit of 0x81f60b8 (len 396) to <ip address>:5060 returned -1: Network is unreachable These messages above exist for each devices that

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]

My server is RH 5.2 and i use Samba 1.9.18p10 and the client use WFW 3.11 with tcp/ip protocols installed. The subnet and the IP Address is right but I still only telnet form WFW 3.11 to Linux...not the reverse. On Linux telnet is working. I've checked it through this cmdline "telnet localhost 23 or telnet 127.0.0.1" . And I also testparm smb.conf and nothing wrong happen. FYI I set

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

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 ? S 09:57 0:00 asterisk -vvvg -c asterisk 29832 0.0 5.1 25924 12208 ? S 11:48 0:00 asterisk -vvvg -c asterisk 31872 0.0 5.1 25924 12208 ? S