similar to: snom subscriptions issue on WRT (2)

Hi, I've just installed asterisk 1.2.1 on my openwrt distro ( I own a WRT54GL by Linksys ) . No problem by now, but I can see that my 3 snom 320, once they started they send subscriptions to asterisk, and I can see that running: sip show subscriptions But, after one hour about, OR when I do asterisk reload , asterisk losts all th snom subscriptions. Someone can help me please? Thanks

I have a preoblem with my snom 320 phones. I have 5 snom phones installed and all of them have 5.2 firmware. All have same settings in the advanced panel. On 2 phones when I press the hold or transfer key nothing happens and * does not start the musiconhold. In the The hold and transfer keys are set as F_R and F_TRANSFER correctly as the others. Other snoms and gxp-2000 work ok. Any ideas?

Hi all, I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for a remote installation. I've got dhcp working and I have provisioning files ready to go. I understand that I need to set bootp option 66 to point to the tftp/ftp/http server. In fact, I have this working completely with the ISC dhcp server The problem is that I don't know how to get

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In

Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other

I thought this can be useful to someone, so here it goes. I am using automatic SysVol replication with the help of "watcher", a recursive incron. https://github.com/splitbrain/Watcher "Watcher is a daemon that watches specified files/folders for changes and fires commands in response to those changes. It is similar to incron, however, configuration uses a simpler to read ini

I upgraded to 0.7.1 from a cvs version from a few weeks before 0.7.1 was relesed. now I am having troubles with my dialing plan and voice mail. As part of the upgrade I re-built the machine so there was a blank slate however after installing 0.7.1 I had no mail box creation script and could not figure out how to go about creating a mailbox, any suggestions would be usefull. I have looked at

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

So, incoming calls on zap work just as I expect them - an intro is played, the caller hits 1 for sale 2 for support or dials an extension. I'm using the privacy option for all extensions. When calls come in from zap, they caller is played the priv-recordintro recording, they say their name, and everything happens normally from there on out. However, when the call comes in from sip and

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Hi Everyone, After a fair amount of faffing ive managed to get the 2000w working with asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls over our BT line). The final solution is to set up outgoing VoIP calls but I now know that without a SIP aware router I can think again! (damn you iptables!) In the mean time I'm trying to figure out why I can't get the

I have a curious problem with an old WRT54GL router, which I use as a WiFi access point on my LAN: Internet->ADSL modem->CentOS-7 computer->WRT54GL router The router has always had a slight problem of losing connection every so often - it used to be every couple of days, but recently it has become much more often. My cure was always to disconnect the power from the router for 10

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

libswfdec/swfdec_as_frame.c | 14 ++ libswfdec/swfdec_as_frame_internal.h | 3 libswfdec/swfdec_as_function.c | 12 -- libswfdec/swfdec_as_interpret.c | 1 libswfdec/swfdec_asbroadcaster.c | 3 test/trace/arguments-5.swf |binary test/trace/arguments-5.swf.trace | 82 +++++++++-------- test/trace/arguments-6.swf |binary

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by

Hi, How does * read and process the extension.conf file?? The reason I ask is that I think it probably has a very large impact on how the calls are routed and processed by the system especially when it comes to least cost routing.. Let me explain...with an example.. I am using the * Devkit to get to grips with the system, so I have and X100P (Zap/1) and and S100U (Zap/2).. Below is my

I have an "outbound-ld" context as follows: [ Context 'outbound-ld' created by 'pbx_config' ] '_1NXXNXXXXXX' => 1. Macro(enumdial|${EXTEN}) [pbx_config] 102. Wait(1) [pbx_config] 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config]

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

On 1.6.0.16-rc1: I'm using app_fax.so to send a fax, and then send a confirm. 'send' => 1. Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config] 2. System(env echo -e "Channel:DAHDI/g0/........\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >${UniqueFile}) [pbx_config] [ Context 'fax-tx' created by

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail