similar to: resync linksys SPA9XX config file from Asterisk

I came up with this a few days ago, mostly used the wiki examples, didn't have time to post on the wiki yet, maybe one of you guys with a few minutes can throw it up there, really, I forgot my logon. http://www.voip-info.org/wiki/index.php?page=Asterisk+Paging+and+Intercom The agi script didn't work for me, wouldn't call the active hint extensions, even though they were there, no

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

> I have a customer that has two buildings that are connected with a > fiber link. We have a single Asterisk server to cover both buildings. > Now the customer went and bought an overhead paging system for the > remote building and they want to integrate it with Asterisk. Is there a > device that can connect over IP or an ATA that has an audio output port? > The buildings

Hi All, Has anyone gotten the polycoms or the linksys phones to accept oprtion 66 on the dhcp request for the address of the tftp config server? We have the dhcp server issuing the proper IP of the tftp server, but the phones just sit there and never try to contact the tftp server for their configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

Hi All, I'm running an AGI, calling a perl script the does number lookups to a remote server. I would like to put a timeout in the script. The problem I'm running into is if the DNS server is not responding, the script hangs and waits for 30 seconds before returning to the Asterisk dialplan. I would like a timeout of 1 second, then return. Here is my clean script:

On Wed, Jan 20, 2010 at 2:42 PM, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I'm using Asterisk 1.4 branch and checking the status of some SIP > Peers with the functions ${SIPPEER(101:status)} and the result is "OK > (48 ms)". ?Seems to work fine. > > Now I would like to use the function CUT to set a variable with the > 'OK'

I installed SPA942 and SPA2101, and experimented with TFTP and HTTP provisioning. It all went smooth for many hours. But then all of a sudden it stopped reading configs from both from TFTP and HTTP. Now I am trying to troubleshoot and cant't find the problem. Once in a while, it does read from TFTP and/or HTTP, but then again, stops reading at all. My other phones, i.e. Grandstream and Aastra

Hi All, Strange issue, recently I started getting a lot of .lock files in the voicemail /INBOX folder preventing proper access to voicemail. I can delete the .lock files and everything is normal. After searching around, I found some SIP lock file stuff but nothing specific to voicemail. Can someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge.

Hi All, http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr Under example: exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) and under description: -userfield: The channel's user specified field. ""-any custom value that you wish to store."" My question is how do you setup more custom fields in the cdr and be

Hi All, Has anyone else noticed that when a sip phone sitting behind a nat registers to asterisk using realtime database, the private IP of the phone is put into the fullcontact field instead of the public contact IP. The database has the correct public IP in the ipaddr field and correct port number in the port field, which is actually what asterisk uses to to contact the device. This

Hi All, I'm running 1.2.9.1 stable. I'm wondering has this patch been applied to stable release or is it still only in CVS. Will this file patch apply correctly to 1.2.9.1 stable? Which file do I patch? I'm guessing app_directory_realtime_1.6.1.patch <http://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch

FYI Kristian. http://www.astlinux.org/ Unable to connect to database server This either means that the username and password information in your settings.php file is incorrect or we can't contact the MySQL database server. This could mean your hosting provider's database server is down. The MySQL error was: Can't connect to local MySQL server through socket

Hi All, I have a Cisco 2600 PRI gateway being hosted on an Asterisk server. The PRI on the cisco is pointing to a customer legacy PBX, the SIP VoIP side of the cisco is pointing to an Asterisk server (1.2.X). In Asterisk, the SIP peer is setup with callerid="some name"<5551212> In a SIP call from the cisco to asterisk, there is no CID name info in SIP debug, so Asterisk

Hi, Is there a way to poll the DND status of a Linksys SPA9xx/5xx phone? The reason I ask is that I'm trying to implement DND + BLF on asterisk. However, the DND softkey on the Linksys phone does not send any feature codes to asterisk. On the flip side, if you disable the Vertical Activation Codes on the phone, then dialing the feature code doesn't display 'Do Not Disturb' on the

Hi All, A couple of customers called complaining that folks were dialing into their PBX trying to use the Directory to locate users, from a Blackberry, and getting frustrated due to the incompatibility of dialing alpha characters on the the qwerty keyboard and not getting through. The issue of course is the Directory application only recognizes numeric digit tones, not alpha characters (not sure

Hi All, Something happened on one of my 1.2.9.1 systems, SIP between * and Cisco Call Manager 4.1, leaving hung or open SIP sessions. No problem now, we found and corrected the problem. But while these hung sessions were increasing to around 480 to 500 sessions, I started getting "too many open files" on the asterisk console and sporadically could not establish new SIP connections.

Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or

Hi All, I'm running * 1.6.0.28 on Debian Lenny. The init'd script starts the asterisk daemon not the safe_asterisk daemon so when asterisk is running and I ssh tot he server then 'asterisk -vr' to attach to the asterisk console there are no colors. If I use the safe_asterisk script to start asterisk, the colors are fine when I attach through SSH. I found this in the init