search for: stevanus

...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

...e 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

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, 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

...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

...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

...co'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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050706/3ab4e07d/attachment.htm

...hem 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

...ter 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

...x8, 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

.../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

...8 0.0 5.1 25924 12232 ? S 13:50 0:00 asterisk -vvvg -c asterisk 5126 0.0 5.1 25932 12240 ? S 13:57 0:00 asterisk -vvvg -c asterisk 6487 0.0 5.1 26016 12336 ? S 14:23 0:00 asterisk -vvvg -c Is this normal? Does anyone experience this? Thanks.. Regards, Stevanus

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=xxxxxxxxxxx type=peer username=0406082250 Regards Anders Svensson -------------- next part -------------- An HTML attachment was scrubbed... URL:

...sterisk picks up and hangs up (though not physically - the caller hears ringing). >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. Stevanus, I think the hanguponpolarity switch is relevant to a patch to to Zaptel that may or may not have actually been added to the released version. I'm not sure. However, thanks for pointing this out -- I've tried it too and didn't get anywhere. >I have similar problems like you. >In...

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.

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....

...tly 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

...arched 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

...en 396) to <ip address>: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

..., 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