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...ing for a way to control another legacy PBX from Asterisk using a CTI interface. Are you aware of any legacy PBX CTI control card that can be controlled by Asterisk? I have an Avaya PBX with CTI interface and researching if I can connect Asterisk to this. :-) Thanks for any hints. -- - David Hajek Daktela - VoipObchod http://www.daktela.com/ http://www.voipobchod.cz/shop/ Tel: +420-226213305 GSM: +420-604352968

Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?

...roxy Auth required. I have loaded Snom with latest 5 firmware. No change. I'm using Asterisk 1.0.9 and have not tried 1.2.X. Looks like this issue is related to http://bugs.digium.com/view.php?id=4801? Does someone get Snom 320 retrieve button working with Asterisk 1.0.9? Thanks, - David Hajek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060122/53bd4d6c/attachment.htm

...ult) (Slaves: 11) Jul 19 17:15:56 ustredna ztcfg: Channel 12: D-channel (Default) (Slaves: 12) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: 12 channels configured. Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna zaptel: Running ztcfg: succeeded Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 hajek@systinet.com http://www.systinet.com

Hello, is it possible Asterisk's ChangeLog will contain a reference to appropriate bug number in bugzilla? This can be very handy. Thanks, David

Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David

...ear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? -- David Hajek http://hajek.net/blog

Hi. samba 4.1.1.. User has unix rights for writing, but samba denies write access to him. On samba server: amistest at samba:~$ id uid=6603(amistest) gid=20(users-nis) groups=20(users-nis),2108(evis),2109(slp),2112(hernie),2126(poj),2133(hto),20000(users) -> user amistest is in "poj" group amistest at samba:~$ ls -ld ACLTEST drwxrwxr-x+ 2 hrubos vema 4096 Nov 27 11:05 ACLTEST

Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some

Hello, I'm looking for IP phones with at least 10 or so speed dial buttons. Can you recommend something which works with Asterisk and does not cost fortune? An option can be analog phone combined with ATA adapter. So hints for good analog phones (EU) are welcomed as well. Thank you, -- David -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, can someone sched a light what exactly mean the read write permissions in manager.conf? [public] secret = private deny=0.0.0.0/0.0.0.0 permit=10.0.0.0/255.255.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Lets say I want some users to use dial through manager interface. But don't want to allow them to run asterisk commands?

Hi, I'm a little bit stuck with Queue app. I'm putting callers into the queue and have them hear music on hold when all (static) agents are busy. This is easy. But when agent become available I want the caller to hear a ringing tone (with message that his call has been routed to the support representative). Is this somehow doable? Thanks, David

...eo codecs configured and session refresh with removed video stream occurs (Reported by Will) * ASTERISK-28049 - res_pjproject build failure (Reported by Jaco Kroon) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by David Hajek) * ASTERISK-28029 - [patch] res_musiconhold : music on hold will not start if previous hold just reached end of file (Reported by Frederic LE FOLL) * ASTERISK-28005 - channel.c: ARI ring only once (Reported by Hajek Michal) * ASTERISK-28032 - Realtime queuemembers a...

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

...ill use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by David Hajek) * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by Joshua C. Colp) * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload (Reported by Sergej Kasumovic) * ASTERISK-28047 - chan_pjsip: Declined video stream is added when no video codecs...

...ill use ports below 10000 (Reported by Joshua C. Colp) * ASTERISK-27854 - rtp: Crash in off-nominal case where RTP instance can't be set up (Reported by Lei Fu) * ASTERISK-28034 - chan_sip unstable with TLS after asterisk start or reloads (Reported by David Hajek) * ASTERISK-28059 - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by Joshua C. Colp) * ASTERISK-27121 - res_pjsip_mwi: Memory leak on reload (Reported by Sergej Kasumovic) * ASTERISK-28047 - chan_pjsip: Declined video stream is added when no video codecs...

...ent of frame format (Reported by ���������) * ASTERISK-24329 - Music On Hold announcement cuts intro of music the first time it is played (Reported by Thomas Frederiksen) * ASTERISK-29081 - res_stasis: Add compare function for bridges moh container (Reported by Hajek Michal) * ASTERISK-29085 - func_curl: Segmentation fault when using CURL after setting httpheader CURLOPT (Reported by P��ter Juh��sz) * ASTERISK-28416 - Unable to get rtp codec payload code for slin (Reported by Brian J. Murrell) New Features made in this release:...

Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David

Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello". I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David

I'm little bit confused which Digium hardware is certificated for use in Europe. It looks like new cards are certificated, like TE4XX series. What about TE110 or TDM400P? Can someone confirm that? Thanks, David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060309/1d1c65a1/attachment.htm