similar to: maximum concurrent ZAP channels .... max conf ports ...

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, 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?

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

Thanks Vamsi I have not been able to locate pwlib - 1.6.6 or openh323 1_13_5. I found the latest versions through sourceforge and I found some older versions on another site, but not these versions. This has been quite frustrating. Anyway, I 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

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:

Hi, I am trying to understand the building of Huffman codes from the code lengths. In the Tremor code first I see that the codewords are being generated by the function _make_words() and then sorted. After this I see some magic code and something related to unused entries. Does the code generate code words for unused entries too? Are these unused entry code words used during the decode

Hey there folks!! I have a question about the maximum number of LANMAN Work Items and concurrent connections from IIS 6.0 to Samba. We have a server for shared windows webhosting running Windows 2003 with IIS 6.0 (with ASP.NET 2.0) connecting to debian 3.1 with Samba 3.0.22 (functioning as a fileserver). At this moment there are about 250 sites running on this server. Now when we make a request

Hi, I remember reading that Asterisk allows only 100 simultaneous calls. Is that correct? If it is so, how is it possible to have a conference call with more then 100 users? I think I read here that some people managed to have 500 people in a conf room... Or, how do I increase this limit? Is it as easy as changing a value in a config.h? Thanks for your help, Emrah

Hi all, In your experience, what is the maximum number of *concurrent* zap channels that you've ever tried with one box of Asterisk open edition? In my case, the max that I've tried was 63 simultaneous connections in a Quad T1/E1 card installed on a Intel Pentium D 3.4Ghz 2GB ram system. Your comments will be really appreciated. Regards, H?ctor. -------------- next part --------------

Hii Everybody, I have a problem with pagination. I have a list of things in my list,like 5,10,20,30 ....pages. If i select the number of items to be displayed,it''s showing like 5 or 10... But after refreshing the page when i was in 10 or 20 pages like that ,it is showing only 5 items per page. Please post the exact code.. Thanks & Regards vamsi -- Posted via

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

How to encrypt and decrypt the url using blowfish in ruby? ex: url= http://localhost:3000?username=vam&paswd=1234&street=hyd&contact=999999999&company=raymarine&city=hyd&state=UP&country=ZP&zip_code=543211 please help its very urgent. Thanks in advance - Vam -- Posted via http://www.ruby-forum.com/. -- You received this message because you are subscribed to

Hi, I'm looking for a reliable provider to use mainly for outgoing calls in the UK, incoming isn't so much of a worry as I think I'm going to accept them over ISDN. Cheers! Steve -- Steve Foy steve@narnian.org

Hi I posted earlier about having issues connecting to Telewest's ISDN, only to find out later Telewest had forgotten to turn it on - hopefully I'm not having a similar silly problem. My PRI span is now up and operational, but when I try to send a call out over it, I just get congestion tones. Occasionally, I get about one second of ring tones, only for it to cut out and play congestion.

Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and

I'm trying to get zap fallback to VoIP working. I dial the zap channel and if it fails I want to then try another route. If the channel is busy (i.e. CHANUNAVAILABLE) it works, however if there's no dial-tone, it doesn't seem to detect this? Using a TDM400 with UK settings. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20

Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3 Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the "lock-up", IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk. Box is debian w/ asterisk built from scratch. My setup is