similar to: iax2 prun realtime peer only can't prune user

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

> I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. > All seems to work normally with realtime voicemail, reads vmbox > parameters from the db fine. When I try to change the password, > asterisk operates normally, "enter new password" ok, "re-enter new > password" ok, "password has been changed" > > There are no entries in

Hi All, I'm using asterisk 1.2.9.1 and mysql 3.23, asterisk add-ons 1.2.3. All seems to work normally with realtime voicemail, reads vmbox parameters from the db fine. When I try to change the password, asterisk operates normally, "enter new password" ok, "re-enter new password" ok, "password has been changed" There are no entries in the mysql.log setting the

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, 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, 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

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'

> -----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 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:

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, Anyone know the sip header to send to a Linksys to resync it's config file? Thanks. JR -- JR Richardson Engineering for the Masses

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 !

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

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

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'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

> 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

Here you go folks: ftp://ftp.ntcp.net/DUNDi_So_Easy.pdf If someone would be so kind as to upload to the wiki, it will be much appriciated. Thank you all who replied to my poll questions. As always, I hope this help. JR -- JR Richardson Engineering for the Masses

We have a couple of positions open, please contact me off-list if interested. http://www.ntegratedsolutions.com/voice-engineer-dallas/ These are full time positions in Dallas, no telecommuters please. Thanks. JR -- JR Richardson Engineering for the Masses Chasing the Azeotrope

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