Re: [asterisk-users] Realtime: Should I say or should I go (now) ?

very good record keeping to be successful. Also on the phone device, the auth name or account name may be 22331 but the display name will be 1001. To make this change, you need a central provisioning server, update the config file and reboot the phone to up

[asterisk-users] resync linksys SPA9XX config file from Asterisk

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 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing li

Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!

ently planning and will hopefully accomplish within Jan-Feb 08. I'm looking forward to the upgrade and having some of the new features 1.4 has like multi threading IAX2. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provid

Re: [asterisk-users] SIP-Realtime and sip reload

with a sip reload? The short answer is, this is how it works, don't reload sip.conf or loose your cache. You can set your phone registration time lower that 3600 so phones re-register quicker. JR -- JR Richardson Engineering for the Masses ___ --B

[asterisk-users] AstLinux WebSite Problem

down. The MySQL error was: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (111). -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- aster

Re: [asterisk-users] Asterisk on Zonbu, impact of transcoding

olume discounts from their suppliers. Zonbu running AstLinux would make for a SOHO SMB SIP PBX market killer. Hm very interesting....says the mad scientist. JR --- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Pr

[asterisk-users] Asterisk & OpenVZ

into play, could use some guidance there as well. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visi

Re: [asterisk-users] Realtime & context

, there is a catch all patch out there is the ether, do a google search for "asterisk alf scherer" and you can catch up on the progress with the patch, try it, it may work for you. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth

Re: [asterisk-users] DUNDI setup help

as well. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Realtime on Asterisk 1.2.24

es_mysql with ARA. 95% of my dial plan is in a database and is working fine for me. Please clarify if you have a moment. Thanks. JR --- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--

[asterisk-users] Backport Func_ODBC question

Hi All, Ingnorant question, how do you apply the backport func_odbc to 1.2 branch? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To

Re: [asterisk-users] Asterisk under VMWare; Great Topic

e shed some more light on why this is and is anyone trying to improve this? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] Asterisk Realtime woes

#x27;t already. www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepa per.pdf Good luck. JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRI

[asterisk-users] Ultrastmonkey? Ultramonkeyast? Astrimonkey? High Availability and Asterisk

, many more who would like to learn or implement this type of solutions. I'm happy to facilitate and document these solutions and share my successes and failures. Please contact me if you would like to participate or just be in the loop. Thanks. JR -- JR Richardson Engine

Re: [asterisk-users] DUNDi, regcontext, softphones.. fail.

> I'm having an issue deploying softphones into my DUNDi/regcontext > setup. My current design is that all SIP users get registered into a > sipregistration context in the sip.conf. I then have a dialplan > function that includes that and does the dial: > > include => sipregistration > exten =>

Re: [asterisk-users] Anyone use the Linksys phones?

ad a directory file yet, wish there was. 10 second boot time, up and running, which is really great. Lots of good features, solid mid to low end cost business phone, customers seem to like it and not many support calls once the users get used to using it. JR

[asterisk-users] Astricon Ride From Airport to Conf Hotel

y take 30-40 minutes. If anyone is coming in around the same time and needs a ride, contact me off-list. Thanks. JR -- JR Richardson Engineering for the Masses ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidt

[asterisk-users] SIPAddHeader cmd from Realtime MySQL, not getting all the 'appdata' field

nfo: sip:\;answer-after=0 accountcode: notes: 1 row in set (0.01 sec) I'm wondering if the colons or the back slash is affecting this coming into asterisk? Thanks. JR -- JR Richardson Engineering for the Masses ___ Sign up now

Re: [asterisk-users] Build your own "appliance" concept

et me know what you think. > http://www.voip-info.org/files/Embedded_Asterisk.doc There are a few tips in here ofr trimming down debian and having a re-producible build environment. Good luck. JR -- JR Richardson Engineering for the Masses ___ Sign

Re: [asterisk-users] Overhead paging over IP

nd device brings the ATA off-hook then bam, your passing audio to the paging system. You can pick these up for ~$200. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asteri

[asterisk-users] DUNDi, So Easy A Caveman Could Do It!

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

[asterisk-users] Quick DUNDi Poll Questions, For All Asterisk Users, Please Give Feedback

ed to follow for new users. Your feedback is appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] Time Limit on Call or Conference Room? "NEW ASTERISK PROVERB"

> On Fri, 3 Aug 2007, JR Richardson wrote: > > > Can anyone point me int he right direction? > >At the risk of coming off in a gratuitiously self-aggrandising manner > quoting myself: > >http://lists.digium.com/pipermail/asterisk-users/2007-May/188438.html >

[asterisk-users] Time Limit on Call or Conference Room?

d outbound calls within the dial plan, per call. I have another customer who wants to offer free calls, for 5-10 minutes with auto disconnect. Can anyone point me int he right direction? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Ban

[asterisk-users] perl script to generate new sip.conf users

Hi All, I remember some folks had put together a web page or perl script to generate many sip.conf entries from a file defining the users, vmbox, secret, CID and other variables. Can someone please point me in the right direction. Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Voicemail .lock- files voicemail box not accessible

someone point me in the right direction to resolve this? I'm runnning 1.2.9 on Debian Sarge. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing li

Re: [asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working, Working now

On 6/26/07, JR Richardson <[EMAIL PROTECTED]> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600.

[asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working

usting the dial-peer dtmf settings in the 2600 and have tried all the dtmf settings in Asterisk. Any guidance will be appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-dig

[asterisk-users] Re: Write to multiple databases as redundancy scheme

setup. Asterisk writes to the Master database and the Master replicates changes to slave databases for backup. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

[asterisk-users] Re: custom cdr fields and cdr_mysql, howto?

On 6/7/07, JR Richardson <[EMAIL PROTECTED]> wrote: 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 ch

[asterisk-users] custom cdr fields and cdr_mysql, howto?

o allow this? I'm running 1.2.9 and addons 1.2.3. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] RE: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

s specifically adapted to your system above you can try adding the command "ulimit -n 8192" to the script that starts Asterisk. -------- JR -- JR Richardson Engineering for the Masses ___

[asterisk-users] RE: Scaling Asterisk: Dual-Core CPUs not yielding gains at high call volumes

nvironments, such as with large call queues. But when you are talking straight call capacity, multiple servers will usually dominate singe servers in relation to cost. All, Nice discussion, and thanks for posting your benchmark results and feedback. JR -- JR Richardson Engineering for the Masses

[asterisk-users] CDR not recording accountcode on SIP Response 302 Call Forward From Phone

r what account the call should be billed to. I have a feeling this is normal behavior for Asterisk as no real channel gets invoked with an accountcode parameter, but there has got to be something that accounts for this situation. Does anyone have a work around or remedy? I'm running 1

[asterisk-users] Re: Polycom or Linksys phones bootp tftp config setup

TP instead if TFTP and the Polycom came right up and acted as expected. I'm still poking around with the Linksys. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[asterisk-users] Polycom or Linksys phones bootp tftp config setup

configs. We can see the proper option going from the dhcp to the phones with ethereal trace. Thanks JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or

[asterisk-users] RE: Ser vs. DUNDi

> With all of the recent talk on the list about DUNDi, I have a question. > From > the outset it appears that SER is often used for high availability > solutions > and as a tool for almost clustering Asterisk boxes behind it. It appears > to > me that DUNDi is providing a lot of this as well. Now I

[asterisk-users] RE: DUNDi configuration problem

> Thank you for the quick response. Do I need to create a route to the > other machine? like a trunk? On the SIP side of things, yes, you can create a SIP trunk for each server-to-server relationship, or you can just send the sip call to the default context and use a goto statement to get the call

[asterisk-users] Re: DUNDi configuration problem

be: priv => dundi-priv-canonical,0,SIP,${NUMBER}@"the real IP Address",nopartial The rest looked ok I think. Good luck. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] Re: FastAGI hangs up channel if server is not available

---- Also you need the asterisk perl agi modules at http://asterisk.gnuinter.net/ Good luck. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] RE: Mr. Spencer Written

> Mr. Spencer written the article "Using DUNDi with a Cluster of Asterisk > Servers " in the > VoIP Magazine and the piece follow: > > [lookupdundi] > exten => _X,1,Goto(${ARG1},1) > switch => DUNDi/priv > exten => i,1,Goto(lookupmysql,${INVA

[asterisk-users] Re: headsets for linksys/sipura phones?

o jacks whatsoever. These work fine with the SPA-942's http://brandcell.stores.yahoo.net/planm1headha.html Plantronics M175, we get them for $23, but this site is cheaper. Regards, JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and

[asterisk-users] Re: is dundi worth pursuing in this situation?

I would use it because I have an ever changing environment and truly need dynamic extension & DID routing between remote locations/servers. Hope this helps. Sorry if it's clear as mud. Good Luck JR -- JR Richardson Engineering for the Masses

[asterisk-users] RE: Voicemail on Different Server (MySQL Replication split thread)

> Having master and slave servers in the same switch fabric is the only > situation in which I would consider replication. > > The cases that I described were with machines in separate subnets. > Replication simply doesn't work that well when there is significant > latency. Did they mention that i

[asterisk-users] Re: Voicemail on Different Server (MySQL Replication split thread)

re there are many stories of failed replication or data corruption when replication is not implemented properly or setup in an environment not particularly suited well for replication. Just wanted to add my own experience. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Re: Voicemail on Different Server, Voicemail with NFS

e wiki page for remote vocemail/mwi. Using NFS works great in a Cluster arrangement, all servers on the same subnet, location. I'll add to the wiki as well. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation prov

[asterisk-users] Re: Voicemail on Different Server, Voicemail with NFS

> -Original Message- > From: JR Richardson [mailto:[EMAIL PROTECTED] > 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 exp

[asterisk-users] Re: agi timeout......clarification

n 4/24/07, JR Richardson <[EMAIL PROTECTED]> wrote: Hi All, Is there a way to specify a time-out option when you call an AGI command from the dialplan? If my AGI fails or doesn't get a response, the call drops, not good. I'm running asterisk 1.2 and calling a fast agi scrip

[asterisk-users] agi timeout

Hi All, Is there a way to specify a time-out option when you call an AGI command from the dialplan? If my AGI fails or doesn't get a response, the call drops, not good. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidt

[asterisk-users] SER/OpenSER, I Finally Get It.............General Observation

foremost, I am still and will always be a true Astriholic; and second, I can't seem to break OpenSER and if you can't break-em, join-em. Can I use OpenSER as a voicemail server, blah, blah, blah??? JR JR Richardson Engineering for the Masses

[asterisk-users] Re: Dial outbound trunk numbers in a round-robin sequence?

> > Hi All, > > > > Customer is requesting 1 incoming toll free #, that dial out to 4 > > different terminating numbers, not ring all at once but ring #1, then > > #2, then #3, then #4, then back to #1 consecutively on inbound calls, > > regardless if someone is on #1. So this is not like a hunt g

[asterisk-users] Dial outbount trunk numbers in a round-robin sequence?

agent queue set with a round-robin sequence. I know agents can do this, not too familiar with agents though. Any ideas on doing an inbound call 'group count' or using agents assigned to dial out instead of ring a SIP extension? Thanks. JR -- JR Richardson Engineering for

[asterisk-users] RE: Using DUNDi in a failover environment

or months, if a registration server fails, I can still call the SIP phone by dipping the database for the full contact info. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users ma

[asterisk-users] Re: Replicating SIP Registrations Across Asterisk Servers

tp://www.astricon.net/files/usa06/Friday-General_Conference/JR_Richardson_Whitepaper.pdf I've been using this setup for months, if a registration server fails, I can still call the SIP phone by dipping the database for the full contact info. JR -- JR Richardson Engineering for the Masses __

[asterisk-users] Re: Lucent TNT - ring timer

cemail boxes to pickup. I found a setting called ringing-timer under sip-options, but it doesn't seem to have any affect. Any ideas? In the SYSTEM profile: set max-dialout-time = 120 I think the default is 20 seconds, increase the time to what you want. JR -- JR Richardson Engine

[asterisk-users] RE: Polycom 501 + Asterisk +Edit buttons

> I have this same feature enabled for one of my clients. You can't > edit the soft buttons on the phone, but you can edit the hard keys. I > remapped a speed dial to an unused hard key. When this key/speed-dial > is pressed, it plays back a sound file that says whether or not night > service is

[asterisk-users] Too Many Open Files, Hung SIP Sessions, Can I Increase File Count?

eased file count. Not sure exactly what this refers to but can someone point me in the right direction? Or am I on the wrong track? Thanks. JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Ea

[asterisk-users] MAX TNT Question

P Info messaging. I'm running 12.0 code on the TNT's Do you have any ideas? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or updat

[asterisk-users] RE: what happened to asterisk wiki???

e. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk SIP to MAX TNT Gateway, Sporadic Echo

m really having trouble isolating anything. I'm wondering if this could be a bad DSP on the TNT, and how would I isolate. We have 600+ DSP's in this chassis. Any experience or ideas with this type of issue would greatly be appreciated. Thanks. JR -- JR Richardson E

[asterisk-users] RE: Polycom reject button

> I have users in my dialplan that go from SIP to Cell > When they are at their desk and they hit reject call, it goes to the > next thing in the dialplan, thus transferring to their cell. Not what > they want. Is it possible to change the reject button to make it go to > voice mail or a new ext?

[asterisk-users] Re: Polycom Firmware

-watch presence, had to shelf the phone and go with a 601 with 1.6.6. That's the only thing I'm aware of is presence seemed to break with the latest firmware. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation

[asterisk-users] Re: Authentication Command

example, * version, hardware, more info about your setup. Make sure you answer the call first, before you invoke the authenticate cmd. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] RE: Asterisk to Cisco's Rescue...again...AuthenticateLD Calls

> From: "Jason Aarons \(US\)" <[EMAIL PROTECTED]> > > Glad to hear you had a workaround. > > I would suggest re-queing your TAC case, perhaps you got a outsourced or > less experienced engineer at Cisco. Their support has varied depending on > which city/group you get. Some have more experience t

[asterisk-users] Asterisk to Cisco's Rescue...again...Authenticate LD Calls

in the first place, but it would have been nice for the Cisco CM and the Cisco Gateway to play nice together. The real hero here is Asterisk, Digium, and the Community that supports it! Thank you All JR JR Richardson Engineering for the Masses

[asterisk-users] requesting real world meetme capacity numbers

user load, perfect audio. I'm working on a conf bridge for 150+ users, could use some advice, if anyone has accomplished such a feat or has any ideas on how. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation p

[asterisk-users] max tnt pri voice channels 56k or 64k, does it matter, selection parameter?

wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.

[asterisk-users] iax.conf setvar= like sip.conf setvar=?

Hi All, I'm running 1.2.9.1, is setvar= implemented in iax.conf in a later version of asterisk? If so, which one? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-

[asterisk-users] iax2 prun realtime peer only can't prune user

Hi All, I'm running 1.2.9.1. I can prune sip realtime peers and users and iax realtime peers but no command to prune iax realtime users. Was this implemented in a later version? Thanks. JR -- JR Richardson Engineering for the M

[asterisk-users] Re: Realtime Voicemail Password Change WORKING NOW

rked. In my effort to be consistent across table names, I did change the voicemail table column 'uniqueid' to 'id', like the sip, iax and exten tables. I changed it back to 'uniqueid' and password update is working fine now. Thanks to all who replied for

[asterisk-users] RE: Realtime Voicemail Password Change Not Working

> Interesting, well if you're seeing the other selects in the mysql.log > then this update not showing up is bizarre. It would also mean that > permissions are irrelevant if doesn't even attempt to change the > password, as you'd rightly pointed out as well. I just tested it again > and this is wha

[asterisk-users] Re: Realtime Voicemail Password Change Not Working

On 1/17/07, JR Richardson <[EMAIL PROTECTED]> wrote: > > 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-users] FW: Realtime Voicemail Password Change Not Working

> 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 ha

[asterisk-users] Realtime Voicemail Password Change Not Working

sword" ok, "password has been changed" There are no entries in the mysql.log setting the new password in the database. How can I isolate between asterisk, realtime driver, and mysql? Thanks. JR -- JR Richardson Engineering for the Masses ___

[asterisk-users] realtime mysql db performance difference with matching extensions

ng 13K records? I'm more inclined to believe the extens matching is the culprit here, not the size of the database. Any thoughts/advice/alternative methods, other experience implementing LCR tables with asterisk? Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Re: Realtime Voicemail Table Column Name Question

ql.c and app_addon_mysql.c. I did not find any reference to "customer_id" so I changed the column name in the database to 'accountcode', so far all is well. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation p

[asterisk-users] Realtime Voicemail Table Column Name Question

OT switch the column name 'customer_id' to 'accountcode' in the voicemail table? Does Asterisk read from the 'customer_id' column for anything? Is the name particular and need to remain 'customer_id' for any reaso

[asterisk-users] Re: asterisk-users Digest, Vol 29, Issue 114

restart the pbx, don't just reload res_musiconhold.so), I could call from pbx1 to pbx2, from pbx2 put the call on hold, pbx2 would play the MOH correctly. Call pbx2 to pbx1, put the call on hold from pbx1 and pbx1 played the MOH correctly. Didn't see any issues past mode=files. -- JR Richa

[asterisk-users] Re: mySQL and to many connections with SQL statement UPDATE

tup realtime access to the database and use the realtime cmd, this will control the mysql access and you won't have to worry about the connections. http://www.voip-info.org/wiki/view/Asterisk+cmd+RealTime -- JR Richardson Engineering for the Masses __

[asterisk-users] RE: Best way to access MySQL data from dial plan

> > > > What is a better way to do it then in terms of performance, security, > and > > flexibility? Using exec and a shell script, or agi or something else? > > Setup extconfig to have realtime access to the database/table you want to pull info from, then in the dialplan use the app Realtime.

[asterisk-users] Cisco Call Manager 4.0 to Asterisk, Anyone have SIP Reinvite working?

h Asterisk, the SIP context shows canreinvite=yes, so should this be working ok, maybe I'm doing something wrong? Thanks JR JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-u

[asterisk-users] Re: Realtime +Mysql +Failover

creases realtime performance on the PBX (reading from localhost than from across a network), scales very well (15-20 slaves for each Master). Also MySQL replication is more mature that MySQL clustering (I'm using 3.23, not 5.0 which is needed for clustering) so the software footprint is small

[asterisk-users] Re: Input on Dundi

ith the last argument (here is have DN_) so when the record is pulled, all the records info is available as a variable like DN_port and DN_ipaddr. This is a really cool command. Hope this helps. -- JR Richardson Engineering for the Masses ___ --Bandwidth a

[asterisk-users] Job Posting, Asterisk Engineer/Sales Engineer, Dallas TX Area

s XP, Microsoft Office, Visio Education: EE/BS/BE, relevant experience and certifications considered. Send resume to [EMAIL PROTECTED] -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-u

[asterisk-users] RE: 0002475: [patch] Allow app_directory to work with REALTIME

th REALTIME 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 and  config.h.patch or  config

[asterisk-users] 0002475: [patch] Allow app_directory to work with REALTIME

://bugs.digium.com/file_download.php?file_id=4915&type=bug> and config.h.patch <http://bugs.digium.com/file_download.php?file_id=4898&type=bug> or config.c.patch <http://bugs.digium.com/file_download.php?file_id=4898&type=bug> . Thanks. JR

[asterisk-users] Re: regcontext, NoOp extension vanishes when extension reload, WORKING

.conf. Let sip module do the rest. Works great. Thanks Guys. JR On 12/5/06, JR Richardson <[EMAIL PROTECTED]> wrote: > > Let me guess: The context in which you have the 2 thru n priorities is > the same one as you're using for regcontext right? > > Don't do tha

[asterisk-users] RE: regcontext, NoOp extension vanishes when extension reload

Let me guess: The context in which you have the 2 thru n priorities is the same one as you're using for regcontext right? Don't do that, bad things will happen (as you've noticed). I'd have to review the code again, but I think what you're seeing is as a result of this. Regards, - Brad No,

[asterisk-users] regcontext, NoOp extension vanishes when extension reload and doesn't come back

e the regcontext function ensure the NoOp priority 1 extension is re-newed each registration cycle, whatever the time parameter is set on. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com --

[asterisk-users] Realtime fullcontact field contains nat device private ip

ct contact info in the sip registration header from the device? Does anyone know how to correct this behavior? It is the same with nat=yes or nat=no. Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Ea

[asterisk-users] Re: DUNDi Asterisk Cluster

ider has to allow re-invites, but only authorized from the dundi servers. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] Re: Asterisk and Max TNT SIP Authentication Issue, WORKING

totally missed the correlation between the CID number translating to a user. The first lesson is to setup the lab to simulate real-world testing. I am curious why Asterisk inturprets the CID number as a user? Thanks. JR -- JR Richardson Engineering for the Masses

[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue, a little closer

regard any user info in the invite. Is there is another mechanism in Asterisk to disregard any user info from an invite? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-u

[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue

g seems to work. I'm wondering is there is a patch that will allow unauthenticated calls in sip? Thanks. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

[asterisk-users] Re: Asterisk and Max TNT PRI to SIP Authentication Issue

what is the sip.conf for 1239 which I'm going to assume is a extension on the TNT Barry JR Richardson wrote: > 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

[asterisk-users] Re: Performance issues in Realtime

rmance, small log file sent from Master to Slave database Setup res_mysql to read from the local Slave database and write to the Master database (Digium Bug Tracker, Mantis Issue 5881) JR -- JR Richardson Engineering for the Masses ___ --Bandwidth a

[asterisk-users] Re: Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

Makefile to allow this? JR On 11/7/06, JR Richardson <[EMAIL PROTECTED]> wrote: 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 r

[asterisk-users] Asterisk and Max TNT Passing calls SIP to PRI, not PRI to SIP Authentication Issue

L PROTECTED] CSeq: 639089 ACK v: SIP/2.0/UDP 10.10.14.131:5060;branch=z9hG4bK006187a112a9899d Max-Forwards: 70 User-Agent: Lucent-Universal-Gateway l: 0 Any guidance will be much appreciated. Thanks. JR -- JR Richardson Engineering for the Masses ___ -

[asterisk-users] Configure Max TNT PRI to SIP with Asterisk

Hi All, Any of you Max TNT Guru's out there have some sample configs for a Max TNT running 11.0.6 code? The example on the wiki was for 10.0 code, it doesn't quite match up with the newer 11.0.6 TOAS release. Any help will be greatly appreciated. Thanks. JR -- JR Richardson Engin

[asterisk-users] Re: asterisk not detecting hangup

> I've enabled those options but it's the same. > > On 10/25/06, Maxi Belino <[EMAIL PROTECTED]> wrote: > > i'm having similar problems (if you find out the solution please post > it) > > > > did you try enabling 'callprogress' or 'busydetect' in zapata.conf ? > > > > Maxi > > > > 2006/10/23, Ark

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