similar to: ztdummy IVR no voice

Hi everyone, I have encountered a hard problem that when i connect my anology phone to channelbank ,I found that i dial a number and create the call,then ,I hangup the call,but ,very quickly,I listen the ringing im my phone,I pick it up ,and found it noting, anybody can tell me this reasons,and how to solve it,Thanks! -- Best regards! jordan pan Location:Shenzhen China

Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid:

hi everybody, Recently,I did a stress test use asterisk . when I concurrent 100 calls to the queue , i found the waiter need very a long time can listen to the music,about 40s and this time, there are about 30 waiters in the queue. everybody can tell me this reasons.whether the queue's ability is limit or the music on hold concurrent is limit . Thanks in advace! -- Best regards!

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug level to 5 and verbose to 5 all

Hi, I have configured samba PDC and BDC servers with ldap backend (debian). All users home directories and shares are on Samba member server (opensolaris). everything works fine except fact that on windows i can't see permissions on folders. On files permissions are displayed correctly. if I tick permissions on folders in windows space they disapper after clicking "apply" but

Hello System: Debian etch with kernel 2.6.18-4-686 or 2.6.18 custom. Asterisk Version: SVN-branch-1.4-r55483M Zaptel Version: SVN-branch-1.4-r2302 modules all ok in compilation time. And modules loaded: ztdummy 5928 0 rtc 13364 1 ztdummy zaptel 181540 1 ztdummy crc_ccitt 3200 1 zaptel In /dev/zap directory I have:

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

Hi all, In sip.conf: [general] regcontext = autoreg [devabc] regexten = 543 creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the dialplan, because there's no device SIP/543. Now I know I can add a line like "exten=> 543,2,Dial(SIP/devabc)" for each and

I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal

hi there i am facing problem in installing the ztdummy module in fedora7, 2.6.22.5-76.fc7 is the version of the kernel. here are some logs for your kind consideration, i have tried varios solution from voip-info.org and many more, but in vain. Message from "/var/log/messages" Sep 13 14:27:14 localhost kernel: Zapata Telephony Interface Registered on major Sep 13 14:27:14 localhost

Hello, I have some ugly numbers given by zttest for ztdummy on an AMD64 box running linux-2.6.15 compiled for Athlon64. linux-2.6.15, zaptel/branches/1.2 r900, jiffies ./zttest Opened pseudo zap interface, measuring accuracy... 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586% 99.975586%

hello, i got this error when i run make after downloading asteirsk from cvs. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYDETECT_MARTIN -fomit-frame-pointer -c -o term.o term.c In file included from

hello i am trying to use res_odbc for sipuser. my connection is working. i have checked using isql. even cdr_odbc is working but i hav problem in res_odbc. i have created user in sip_buddies table but asterisk is no getting user from this sip_buddies table. /etc/asterisk/extconfig.conf [settings] sipusers=>odbc,asterisk,sip_buddies sippeers=>odbc,asterisk,sip_buddies

Question is quite easy: How am I supposed to configure Asteirsk to have the same extension, in 2 differents domains. In the general section of sip.conf, I add the domains, But how to say to Asterisk : user1@domain1 > Pasword1 user2@domain2 > Pasword2 Thanks for your help !!!!! JM -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks

I have read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- ------------------------------------------------------------ Erick

http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even

Civilization IV Beyond the Sword expansion. I'm using Wine 1.1.44 with the latest intel drivers (2.12.0) and mesa at 7.8.2. OS is GNU/Linux, Gentoo distribution Both the minimap at the lower left corner and leaderheads don't show up, showing just a black section on the screen instead. Also, some artifacts are showing up in forests and jungles. This is very different from the results