similar to: BT-100 Transfer!!

can anybody identify why the CLI is issuing a failure message while debug shows everything is fine???? this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 -- SIP Seeding '52221' at 52221@192.168.70.26:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]:

Has anybody seen this behaviour? sip conf is stored in mysql database in 2 tables ast_config for static (general) key/values sip_buddies for sip extension detail. database on the same machine as asterisk Grandstream phones (I happen to have 2) register with asterisk via sip with accounts and passwords successfully for a variable period of time. Then after a while, i get errors which appear to

Apparently asterisk cannot reboot gracefully (unattended) when using realtime.... MySQL RealTime: Failed to connect database server asterisk on 127.0.0.1. Check debug for more info. WARNING[3763]: MySQL RealTime: Couldn't establish connection. Check debug. Since asterisk starts before mysql it will never start any workaround for this? g Regards Greg Cirino

It's been hours since I've seen a post from this list Must be broken again. Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You

I am trying to resolve a problem where grandstream phones (only) fail to register after a period of time. I have a mysql realtime setup that appears to work, but fails for no reason. before i classify the realtime system as unusable, I am trying to isolate the problem. One thing I have noticed is the odbc driver continues to load even though I have removed (from what I can tell) references to

Has anybody been able to get in touch with anybody at digium today? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just

does anybody have an idea what the difference and significance of sip seeding and registration is. g Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX

from voip-info wiki Asterisk automatic daily restart To automatically restart Asterisk you can add something like this to cron # Restart Asterisk PBX once a day to prevent any problems from piling up 10 7 * * * root /usr/sbin/asterisk -rx "restart now" >/dev/null 2>&1 or 10 7 * * * root /usr/sbin/asterisk -r -x "restart gracefully" >/dev/null 2>&1

it appears the cvs for zaptel as of 12/13/04 am is missing at least 1 file -- wcfxs.c greg Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development & Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When

Good day all How do i set a connection between 2 asterisk servers via PRI In Bri I would set one to NT and TE How shoud the zapata.conf and zaptel.conf look And how should the cable be? All I got on the web was to set one to "pri_net"...this cant be all? And the cable > pin1 <--> pin4> pin2 <--> pin5> pin3 <--> pin6> pin4 <--> pin1> pin5 <-->

Hi, Has anybody had any problems getting digium hardware lately? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a

Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext => 700 ; What ext. to dial to park parkpos =>

Hi My card is working, but when I reboot the machine, most of the times it is not working, I get "ztcfg: ZT_CHANCONFIG failed on channel 1: No such device or address (6)" To make it work again I have to shut down, remove the card, reboot so kudzu will remove the config. shut down again, put the card back in, reboot, now kudzu see it, I choose "Ignore" and then it's

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

from the asterisk messages log: Registration from '<sip:40852@192.168.70.2>' failed for '192.168.70.25' the only place I can see extension 40852 linked to the ip is in the phone's configuration. sip.conf [40852] ;for a grandstream bt100 musicclass=homeline pedantic=yes accountcode = 40852 amaflags = billing ;callgroup = callerid = 40852 <40852> canreinvite = no

iH i have an IAXy which i can make calls from but am unable to call. when i dial the extension assigned, i get the following from the console; -- Executing Dial("SIP/5801-b665", "IAX2/5899@192.168.0.5") in new stack -- Called 5899@192.168.0.5 -- Call accepted by 192.168.0.5 (format ULAW) Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected

Hi again. I cant get my Budgetone registered in Asterisk, and I cant find what's wrong... uff. This is my config: This fragment is from my sip.conf: [12345] type=user user=12345 username=12345 secret=12345 authuser=12345 qualify=1000 nat=no host=dynamic dtmfmode=rfc2833 reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw allow=alaw context=sip_default And this is from my

Does anyone know what this means "Warning [65542]: chan_sip:c:501 retrans_plct: Maximum retries exceeded on call 7438737dc873850@172.16.0.52 for seqno102 (Non-critical Request. 172.16.0.52 is the Asterisk Server I'm guessing that I have something miss configured just not sure what it is. If you need more info just ask.

I hope this is an appropriate question for the list.. I am looking for a VOIP termination provider who can offer the following: -Flat Rate DID's in lots of areas -GOOD customer service/support with quick response times -Toll Free DID's at a reasonable rate -Reliable/Redundant network and availability etc. So far I have tested 4 providers which I will not mention here. I have found two

I know I'm getting tired of looking at code, but why is the function register_verify defined in 2 different files? chan_iax2.c line 3860 static int register_verify(int callno, struct sockaddr_in *sin, struct iax_ies *ies) chan_sip.c line 4869 /*--- register_verify: Verify registration of user */ static int register_verify(struct sip_pvt *p, struct sockaddr_in *sin, struct sip_request *req,