similar to: Test System?

I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded

Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones?

I think I have a bad FXS module on my TDM400P. Channel map: >> >> Channel 01: FXO Kewlstart (Default) (Slaves: 01) >> Channel 02: FXS Kewlstart (Default) (Slaves: 02) >> >> 2 channels configured. >> ZT_CHANCONFIG failed on channel 1: No such device or address (6) The zaptel loads ok, but the wctdm reports No such device or address and stays loaded. The

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

Anyone else ever have the problem of asterisk picking up with voicemail after one ring on an extension? I'm using free world dial up's IAX2 service, and I can make calls but received calls get a voicemail pickup after one ring. No decent answer on google, cannot find anything that seems to be wrong after comparison with a working config. Using: latest CVS-HEAD (resnatched and built

Hi List; Why I need zlib1g to do installation for Zaptel? Will zlib1g do compression or it will what extactly do during the installation process? Regards Bilal ____________________________________________________________________________________ Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/

Hi, > I've never run against a commercial PBX that didn't need > maintenance. Acknowledged. > VM hard > drives fail, > ... > Asterisk is > every bit as stable > as the old-gen KSUs and PBXs. There are big differences. As I know of no other PBX that uses 'consumer' hardware, asterisk is also struggling with problems in the underlying Hardware. And

I have all my SIP users in a realtime database. I would like to use MySQL command to query the database and use the results from the query to page all the phones found in the query. The results from the MySQL query will be multiple rows of extension: Something like: mysql> Select extension from sip where extension like '6%' 6001 6002 6003 ex.... I need to put all the results into a

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Hi all: I've got some core dumps when I use chan_h323. I dial an extension using h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes * hangs, sometimes not. The client used for test es SjPhone (http://www.sjlabs.com/). This is the data for one core dump: (gdb) bt #0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790 #1 0x41f8879c in create_connection

Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks...

Anyone have any suggestions on free sip phone software for windows?? Only have one IP phone and want to have one other computer hooked up to my Asterisk box for testing.

Hello! Can anyone recommend a good/handy/nice sip soft phone? I have already done some testing with kphone and gnome meeting (which cant do sip).Can you recommend a open source project? It should mainly be practial and have a address book. I found kphone quite unstable, the address book is designed quite poor, and if you would like to transfer a call with the transfer button you cant access the

Hi list, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050116/f75bd2fe/attachment.htm

Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens) an register it with asterisk? any suggestions? thx in advance. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

Hello! I'm searching for a way to call ATA (IAX or SIP) that is not registered with any server or proxy. Is it possible to make such a call from a softphone to an ATA just with IP? Something like (sip:// or iax://)1111@210.12.34.45 (where 210.12.34.45 is ATA's public ip)? Regards, CuPoTKa.