similar to: Asterisk & MyPhoneCompany.com (aka Talk(n))

I am trying to get Asterisks to connect to our StanaPhone so that I can use it to route my outgoing PSTN calls to. We have a free account and if I can get this working are willing to pay for an actual minutes with them. Here is what I have in my sip.conf: [stanaphone] type=friend secret=pAsSwOrD ; skewed for this message. username=3475341914 host=sip.stanaphone.com

I want to test outgoing VoIP Cheap, and they have a $5.95/month plan that would be perfect. I want it to interface through asterisk though, and I heard some providers are blocking asterisk. I just want to make sure that it still works before I sign up. Also, if you have any experience with them, let me know. Thanks, -Todd

I have really tried to do my best googling and wiki-reading before asking this question. I couldn't find the answers there so I throw myself at the mercy of the list... I get excellent quality for SIP -> PSTN and PSTN -> SIP calls, however when I or anyone else calls from PSTN -> * the voice menus are oftentimes very choppy. Sometimes they are absolutely perfect and I cannot tell

I have an idea, tell me if this wouldn't work... I know it's really ugly, but it might help some people until we can get round robin DNS checks for peers... Since * does not do GetHostByName() again until you reload your config, and BroadVoice and I'm sure other sip providers are using round robin DNS, why not create 2 [<your server here>-out] contexts in sip.conf, and then in

CVS 7/16/04 (the latest one I have b4 today) seems to have this problem too... Anyone else having problems with the current CVS ignoring calls after about 5 minutes of being up? I've also noticed that no matter what I set default_expiry to in sip.conf, it starts at that number and then jumps to 44 seconds... not sure where it's getting the number 44 from, it seems to use that all the

I know this must have been asked before, but I was just wondering, the manual says it can do attended transfers, has anyone gotten this to work successfully? How did they do it? Is it possible to do attended transfers with the 'T' dial option? If so, how? -Chris Chris Shaw IS Manager Water Tech Industries Phone: (888)-254-8412 Fax: (503)-261-9118 E-Mail: chriss@watertech.com

I am having problems getting the latest CVS right now. A cvs checkout asterisk -t gets to this part and sits forever: S-> server_register(fpm-world-mix.mp3, 1.1, , , , , ) S-> Register(fpm-world-mix.mp3, 1.1, , , ) Anyone know how I can just skip the file? Travis Conway EFS, Inc. Information Technology Desk:?? (334) 215-6551 Mobile: (334) 391-4450 mailto:travis@homeoffice.quikpawn.com

> -----Original Message----- > From: Chris Shaw [mailto:chriss@watertech.com] > Sent: September 7, 2004 4:40 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 > w/ojitterbuffer enabled? > {clip} > > If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP >

I have a bridge that has gigabit interfaces. The machine in question has the fun job of being a Bridge, Firewall and SMB server. Both of the Gigabit interfaces are connected to workstations directly via Xover cable (well MDI-X to be exact). My question is, if I enable jumbo frames on the gigabit interfaces will that make any difference in overall transfer rate of the bridge? I was thinking it

Ok I have googled and googled and combed through the wiki for an answer to this and have come up empty. What I'm finding is that when a user changes their VM password, it is saved somewhere like maybe the CSV database or something because when you log in, the new password works fine, but it's not saving to voicemail.conf. So new passwords are lost when asterisk is restarted and people

Does anyone have any useful contact information for Broadvoice? I've been unable to call Argentina from my Broadvoice line for almost a month now. Their techs keep telling me they're "looking into it" but I get no real information. I've had several other tickets open for almost 8 months now. I called their PR company this morning and apparently they dropped Broadvoice

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Hi, I'm trying to use transform my mercator locations into utm but I'm doing something wrong because only x is transformed ... see > xy.sp SpatialPoints: loni lati [1,] -8.85 38.16 [2,] -9.19 37.99 [3,] -9.11 37.97 [4,] -9.06 38.15 [5,] -9.03 37.87 [6,] -9.14 37.81 [7,] -9.09 37.70 [8,] -8.95 37.45 [9,] -9.17 37.37 [10,] -9.00 37.34 Coordinate Reference

Dear all, I have tried a lot of things to make broadvoice work with asterisk , but I failed each time. Please suggest a good service providers that I can use with asterisk for outbound and inbound calls. -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal

I was alerted the other day by of all people, my mom, that she wasn't hearing a "ring" when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, "another damned Broadvoice issue." For kicks, I upgraded to 1.2.6 today, and the end

Trying to update to the latest cvs, but Asterisk complained that zaptel was too old. Updating zaptel gives me this during the make. Any ideas, the searches and Wiki gives me no hints. cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Hi Last week I installed Asterisk (release1) with digium t100p single span T1 (wct1xxp) board on Dell GX270 pc configured for PRI. Asterisk/t100p is currently the only user of the t1 line. All worked well for about a half a day, PSTN to SIP phones to non-SIP IP phones etc. Alas, since then I consistently get multitudes of blue alarms on all b-channels followed by a loss of d-channel: Aug 31

don't quote me on this but I believe the earlier assumtion is correction. I think you need to have RTP going bothways otherwise the call will disconnect. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Shaw Sent: Friday, August 13, 2004 12:40 PM To: asterisk-users@lists.digium.com Subject: Re:

Hello, i try to setup a linux laptop for homeoffice with login for ad users. The last few days it work like expected. today wbinfo -u return no user, getent passwd <username> also. wbinfo -a "SAMDOM\user" Enter SAMDOM\user's password: plaintext password authentication succeeded Enter SAMDOM\user's password: challenge/response password authentication succeeded wbinfo -D