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Hi, I'm trying to install asterisk on Dell power edge 2800 running Fedora core 3. I don't have have any zaptel cards, so trying to use ztdummy. /dev/zap is successfuly created... but I see some problems while starting asterisk ... chan_zap fails to load. Can somebody please help me in overcoming this problem. I was able to run asterisk on other normal PCs running Fedora core 3. Is this

Hi All, Is it possible to go beyond 250 concurrent ZAP channels with some tweaking or workaround ? Meetme uses zap channels, so we could have a max of 250 conference ports. Is it possible to higher this ? "An Asterisk system can only handle a *max. of 250 concurrent ZAP channels*. This is due to the design limit (255) within the ZAP channel driver." Thanks, ~Vamsi -------------- next

Hi, Realtime SIP and Extensions are working fine but facing some problems with Voicemail. Added an entry to extconfig.conf voicemail => mysql,asterisk,voicemail_users Created the corresponding table and an entry for mailbox 201. This is also reflected in the CLI as shown below. CLI> realtime load voicemail mailbox 201 Column Name Column Value

Hi, I was able to work out SIP trunk between Asterisk and CCM 4.x without any issues. Whereas SIP trunk in CCM 5.x is not working with Asterisk. Asterisk is sending OPTIONS messages to CCM 5.x for which CCM is not replying. For the same reason Asterisk is marking it as UNREACHABLE. Anybody got Asterisk and CCM 5.x intergation working. How can I fix the problem which I'm facing with CCM 5.x?

If your phones are setup to connect to the asterisk box by name, then a smart DNS server can just point phones to the backup box after failure. However, since asterisk running on the backup box doesn't know about the phones, this is only half the solution ________________________________ From: Mohamed A. Gombolaty [mailto:mgombolaty@noorgroup.net] Sent: Thursday, June 30, 2005 8:30 AM To:

Hello, I have Cisco CallManager 3.3.4 and Asterisk@Home latest version. I have earlier tried getting Asterisk to register with CCM via H323 and failed. Back then, I learned that this is a known bug in Asterisk. Also people who tried doing that had also succeeded in getting calls to go through only one direction like from CCM to Asterisk. I am not that expert so excuse my ignorance with this

...think by using the asterisk-oh323 branch under channels in the asterisk source tree I will have more luck. At present it seems to compile successfully, but fails linking due to a lib expat, which I have no idea where that comes from. Regards Mark Date: Tue, 8 Mar 2005 13:18:30 +0530 From: Vamsi Pottangi <vamsipottangi@gmail.com> Subject: Re: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <cd10de77050307234860248b41@...

Hi, Case1: --------- --> extensions.conf exten => 1023,1,Voicemail(101) exten => 1023/101,1,MeetMe(200) Case2: --------- -> extensions table (using realtime extensions) +----+---------+----------+--------+----------+---------+ | id | context | exten |priority| app | appdata | +----+---------+----------+--------+----------+---------+ | 29 | default | 1023 | 1 |

Hi All, I've installed asterisk and manually configured IAX/SIP users. Everything works fine, I'm able to call other extensions. But when I installed AMP and created new extensions, I'm not able to call those extensions. I get the message that the extension is busy and it is forwarded to voicemail. What am I missing here? The workaround I found is by modifying the