similar to: Getting 488's between two c7940's

Hi Everyone, After a fair amount of faffing ive managed to get the 2000w working with asterisk for IP -> PSTN calls (i.e. get the phone to make and receive calls over our BT line). The final solution is to set up outgoing VoIP calls but I now know that without a SIP aware router I can think again! (damn you iptables!) In the mean time I'm trying to figure out why I can't get the

I want Asterisk to delay answering the POTS line via a Wildcard (a Zap channel) by some period of time, either a number of rings or just a number of seconds. I have tried this: [from-pots] exten => s,1,Wait(30) exten => s,n,Answer ... exten => s,n,Dial(SIP/brian&SIP/joe,10,H) exten => s,n,Voicemail(u2001) exten => s,n,Hangup exten => s,103,Voicemail(u2001) exten =>

For the last, I dunno, week or so I've been unable to build R... I finally took the time to investigate and the crash is in building 'methods': all.R is unchanged ../../../library/methods/man/methods.Rd is unchanged make[2]: `Makedeps' is up to date. ../../../../library/methods/libs/methods.so is unchanged make[1]: Nothing to be done for `Rfiles'. Any clues to where this

we are trying to adapt the configure in order to work with latest gcc3.3 (from apple) and g77 3.4 (from http://gravity.psu.edu/~khanna/hpc.html ) At the moment there is no need to define the __DEBUGGING__ but there is still a problem with the -lcc_dynamic does any of you know how to check for this library and explain why we see -lcc_dynamic often passed as an ld flag. Where to find doc on

http://lists.digium.com/pipermail/asterisk-users/2005-September/125146.html -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com genom Brian J. Murrell Skickat: to 2006-02-02 20:14 Till: asterisk-users@lists.digium.com ?mne: [Asterisk-Users] delaying "answer" for a number of rings or anamount of time I want Asterisk to delay answering the POTS line via a

Hi All, After configuring and compiling OpenSSH version 4.6p1 in my SUN Solaris8 ultrasparc machine, I run "make tests" and got the following errors. The compiler used is GCC3.3. run test sftp.sh ... test basic sftp put/get: buffer_size 5 num_requests 1 sftp failed with 1 test basic sftp put/get: buffer_size 5 num_requests 2 sftp failed with 1 test basic sftp put/get: buffer_size 5

Hi all, I have the Asterisk set up and 2 softphone (Xlite) set up on two other PC. With the following configuration, I can use one softphone (2000) to call the other one (2001) and/or the voicemail at 2999. Here is my problem: 1. When I dial 9+nxxx-xxxx with one of the softphone to the PSTN via X100P card, I got busy tone. (i.e. I want to use the phone line which is connected to the

Has this been solved? When I park a call, the caller hears a second of music on hold and then the whole system crashes. I can restart with a simple (asterisk -cvvv), I don't have to reboot or anything John

When Asterisk first starts up, it will attempt to "bring up" the B channels on any PRI circuits. If you are using A@H then you can log on to the Asterisk CLI (asterisk -r) and then do "stop now" to stop asterisk. Start up Asterisk again by typing asterisk -cvvv at the Linux command line. You should see a bunch of messages on the terminal and then you'll get the Asterisk

Hi List; I am using Firefly softphone Version 1.9.9 Build 4521 and I select IAX protocol and did the configuration in Network1 (and I checked the Active checkbox) as following: Server: 192.168.8.4 username: iax2user1 password: password In the Asterisk, I did the following configuration on the /etc/asterisk/iax.conf: [iax2user1] type=friend context=internal username=iax2user1 secret=password

Hello; I am installing asterisk and dahdi on ubuntu "and I used my username bghayad to login for ubuntu and do the installation, actually I feel my problem is related to the?username and permission but I am not able how to fix it", I am facing now mainly the following two problems: The first one, asterisk is not starting automatically although I did sudo make config (for asterisk and

Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-1.4.22.1 All works fine, dahdi status is: asterik:/data/programmi# /etc/init.d/dahdi status ### Span 1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER) asterik:/data/programmi#

Hi; I have entries as below in DB, mysql> select * from sip_buddies; +----+------+----------+------------+---------+------------+--------+------- -----+------------+----------+------+ | id | name | context | defaultip | host | mailbox | type | regseconds | ipaddr | username | port | +----+------+----------+------------+---------+------------+--------+-------

I installed the TDM400P and installed it on a system running on Fedora Core1, please refer to the steps below. I connected an analogue phone to the FXS port. When I pick up the speaker I don't hear the dialtone although when I press a number key I hear a DTMF generated. What could be the problem? I appreciate your help 1- First I installed the TDM400P in the PCI slot and connected

Well in my setup I have a few IP phones connected to Asterisk as well as POTS phones on my analog line. When a call for my daughter comes in on the analog line (determined from callerID) I send it to her own voicemail after 20 seconds of ringing. It all works quite well. Here's a step-by-step of what happens below: 1 - a call comes in and Asterisk rings SIP/Brian and SIP/joe for 30 seconds.

Hi all, Any one tried installing Asterisk on Linksys WRT54G? We have but facing problems with SIP to SIP calls. The phones ring and calls are established but we cannot hear any voice at all. I tried allow=all in the general section but did not work. So I forced ulaw. Can any one please check it out and let me know what is wrong? Here are the conf files: Asterisk Version: Asterisk

Hi, I configured asterisk on redhat linux 9 box. I installed two different ip softphones (SJPHONE and X-PRO) and got them registered with asterisk. The call from one phone to another does get routed via asterisk, but there is one problem coming up. As soon as call is accepted by the end user , it is automatically disconnected with the error "cannot align media streams". If I enable SIP

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

No, it will dial like a pass-through simultaneously to sip/iax extensions. If you were to dial out to an analog port though, that would be different. So in essence, you can have all the phones ringing at the same time. Greg -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian J. Murrell Sent: Thursday,

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24