similar to: Outgoing busy

As I got to compile 1.0.6 and got it to run but having the same problem as before I thought of creating a new mail thread about this instead of continuing with one where topic is about something else. (Sorry) So, I can't do register anymore. It worked just a couple of days before and I haven't done anything special as far as I remember. *CLI outputs: Feb 28 22:16:55 NOTICE[53475]:

First of all. Asterisk was not functioning very well lately as I couldn't register. Output from *CLI: knivby*CLI> sip reload Feb 28 21:17:22 WARNING[143771648]: chan_sip.c:1310 create_addr: No such host: ipkund1.rixtelecom.se Strange, because I can ping the host. Have also tried changing the hostname for its IP address but no change in result. Output the same but the ipkund1 changed to

Hi! Is there a simple way for an * newbie to force * to use different sip-trunks for different calls. I have 2 siptrunks, one for inland calls and one for international calls. All in country numbers starts with 0 and all international starts with 00. This I have configured in the outbound routing. But * always use the incountry trunk because the 0. dialpattern is also true for international

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

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, 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! Probably another newbie question. Is it possible to run * on one processor and MySql on the other in a double cpu server? Anders -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/6e3590b5/attachment.htm

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, 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 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, 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, 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 ...

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received

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

I have tried to update openssh-3.1p1 of our system that uses RH7.2 (Scyld). I is pretty much a standard Redhat 7.2 install with openssl-0.9.6b, zlib-1.1.4 etc. I have gotten openssh to work after some initial issues, but I still have not been able to get openssh/sshd to work with tcp-wrappers. I have in hosts.deny ALL: ALL: and in hosts.allow ALL: localhost, 127.0.0.1, 192.168.1. and still I

Hi. I'm using ubuntu 10.04 and I'm in the process of buying a ups. This is it (eaton 5115 ups) with usb interface http://powerquality.eaton.com/05146561-5591.aspx?CX=3 I had a look at the ups compatibility chart here: http://www.networkupstools.org/stable-hcl.html But I can't find this particular ups listed. I plan to use NUT v. 2.4.3 that I can get from the

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 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, 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