Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850

uname -a Linux xxx 2.6.18-1.2798.fc6 #1 SMP Mon Oct 16 14:37:32 EDT 2006 i686 i686 i386 GNU/Linux modinfo zaptel | grep ^version: version:1.4.0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

Re: [asterisk-users] zaptel 1.4 on fedora core 6 with dell pe 2850

I can run zaptel 1.4 normally in other machine on the same OS, only can't run it on 2850. It hangs the OS.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] web-meetme cbmysql not registered

HI, today I download Web-MeetMe-3.0.0 for asterisk 1.4.0 but when I call the extension which invoke cbmysql, a warning appears: WARNING[20225] pbx.c: No application 'CBMysql' for extension (default, 1995, 3) I check the application, it didn't registered CLI core show

[asterisk-users] Newbie question: How to config rtp packetization in 1.4?

Hi, any one test rtp packetization in 1.4?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] asterisk 1.4.0 didn't compile chan_zap.so

HI, I use fc6 , the latest stable asterisk 1.4, zaptel 1.4 and libpri 1.4 after I installed zaptel and libpri. I can start zaptel. and my te410p card got green lamp. but when I continue to compile and install asterisk, I can't find chan_zap.so compiled. and in my asterisk cli. I can't 'help

[asterisk-users] Can I disable send e-mail feature in the voicemail application?

HI, all Can I disable send e-mail feature in the voicemail application?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Re: Asterisk + Huawei

need debug * and Huawei, not * and client___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk + Huawei

Can you give more debug information. Usually codec incorrupt can cause failure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] real time billing system

a2billing___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTCP and RTP packetization in 1.4

Dose this trunk do just like IAX2 trunk, to reduce bandwidth?___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTCP and RTP packetization in 1.4

Okay, Thank you. So packetization is a feature of RTP and can work with all of the codecs, isn't it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Iax Netstat Output

I know what, if I use ZAP-IAX2 ---IAX2, I also got one direction poor. But if I use SIP-IAX2 ---IAX2-, every think is OK. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] asterisk, iaxmodem, hylafax quality problem

Are you use digium card? digium pri card offen cause many problems, check zttest___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] IAX2 register refuse but Dial cmd works!

Hi, I just set two asterisk connect with iax2 trunk. B server [user1] type=user trunk=yes context=from-trunk username=user1 auth=plaintext secret=passwd notransfer=yes A server register = user1:[EMAIL PROTECTED] I notice on A's CLI, it shows Registration of 'user1' rejected: 'Registration

[asterisk-users] iax2 trunk call limits

Hi, all Can I limit calls in one iax2 trunk just like sip peers do? How? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] How to setup multiple iax2 trunks between two asterisk server?

Hi, all How to setup multiple iax2 trunks between two asterisk server? thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] IAX2 trunking

I can take 30 calls in one trunk with good voice quality more calls cause awesome sounds___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] IAX2 trunk voice quality: how many calls cause jitter?

Hi, I use IAX2 trunk between two asterisk server. At a few calls (less than 30) enviorment, both caller and callee hear each other clearly. But when calls reach 45 or above, the quality of sounds is bad. I wonder if a IAX2 Trunk should limit concurrent calls? I use ILBC codec in the trunk.

[asterisk-users] iax2 warning!

I always got this warning after I'm using IAX2 channels . Sep 11 21:44:09 WARNING[30229]: chan_iax2.c:6536 socket_read: Received trunked frame before first full voice frame What's it mean?___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Re: the sounds quality of IAX2 channels are notgood as SIP channels?

I use the latest version of zaptel, asterisk___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] the sounds quality of IAX2 channels are not good as SIP channels?

I use both IAX2 channels and SIP channels. IAX2 channels reduce bandwidth effectively. But sometime my cli show NOTICE[1281]: chan_iax2.c:1628 iax2_destroy: Avoiding IAX destroy deadlock WARNING[1281]: chan_iax2.c:708 jb_warning_output: Resyncing the jb. last_delay 28, this delay 1227,

[asterisk-users] Re:sip giving problems, please help.

Yes, I also get these problems occasionally Sep 4 17:44:49 WARNING[1365]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:44:49 WARNING[1364]: channel.c:787 channel_find_locked: Avoided deadlock for '0x8224468', 10 retries! Sep 4 17:52:15

[asterisk-users] Newzealand zaptel DTMF problem

Hi, all. I'm using TE411P card and asterisk in Newzealand as a VOIP gateway. At the begining, all works fine. I have 50 concurrent calls in busy time. But recently I found some users can't send DTMFcorrectly to my gateway. I found some of them sent less DTMF digits than they acturely dialed,

Re: [Asterisk-Users] TE410P and SPANDSP

TE405p and spandsp works good. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] TE410P and SPANDSP

Hi, All Does any one has successful experience use te410p and spandsp together? Could they work well with all 120 channels receive/send fax at the same time? My practice is that rxfax always get broken fax page. Help!___ --Bandwidth and

Re: [Asterisk-Users] TE410P and SPANDSP

- From: Ma Zhiyong [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 24, 2005 9:21 AM Subject: [Asterisk-Users] TE410P and SPANDSP Hi, All Does any one has successful experience use te410p and spandsp

Re: [Asterisk-Users] TE410P + SPANDSP fax problem

 Hi, Steve Thank you for your hard work. Yes, I use EuroISDN. My four E1s connect toan Alcatel S12 Switch that works on PSTN. My Telco. turn CRC4 off, sodo I. And I'll askthem turn it on. I have no X-Windows problem. I run * on Redhat 9.0 and my run level is 3. That

Re: [Asterisk-Users] TE410P + SPANDSP fax problem

I'm sure use my telco as clock src, and my libtiff is also v3.5.7, while problem still exist. Shall I contact with my telco for timing? in zaptel.conf, I set span=1,1,0,ccs,hdb3span=2,2,0,ccs,hdb3span=3,0,0,ccs,hdb3 span=4,0,0,ccs,hdb3 To trace rxfax, just turn on debug trace

[Asterisk-Users] TE410P + SPANDSP fax problem

Hi, I just setup a fax server by spandsp. But it doesn't look good. Becauseeach fax I received from my fax machine is not completed. I use te410p work with it. While the voice call is good. Any ideas? Trace shows that the fax is received successfully. Aug 17 12:01:10

[Asterisk-Users] te405p and dell poweredge

Hi, I want to install * and te405p on Dell Poweredge 1850. Can I do that successfully? Any one has successful experience on that scenario? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] spandsp configuration

Hi, I receive fax using spandsp. It works, however the tif file it stored has no good quality. Any method to configure that? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] digium card and fax

Hi, I practiced fax transmission over digium card and spandsp. I send the A4 page from TDM FXS port and received it by spandsp. Whole page was transmitted and stored in a tif file. When I open the file, I get a horizontal A4 page and theimage on the pagewas flatted. Any one has idea?

Re: [Asterisk-Users] spandsp

It sure works when I createthe diretroy "/var/spool/asterisk/fax/2201001/".HoweverI found it coundn't received the whole fax which eicon card sent. That is, when eicon card send a page with 30 lines text, spandsp only received18 lines. Then hung up. -- Goto

[Asterisk-Users] spandsp

Hi, I installed spandsp and test it with Eicon card. When fax begin from eicon card to spandsp. It fails and shows: -- Executing RxFAX("Zap/124-1", "/var/spool/asterisk/fax/2201001/1115604630.5.tif") in new stack -- Channel 0/31, span 4 got hangup May 9 10:10:41 DEBUG[4967]:

[Asterisk-Users] CID Number problem

Hi, all. I'm glad I put asterisk and hylafax togetherjust like PSTN-Asterisk-Hylafax-Email.And the fax2email functionworks well. But I also find some bugs about CID number. I use TE405P as gateway and Eicon PRI card as fax card. When I receive the caller number from PSTN, I

[Asterisk-Users] ast-rad-acc.pl problem

Hi, All I installed a PortaOne's Radius client for my asterisk Server. But I can't run ast-rad-acc.pl after installation. It says "Can't call method "val" on an undefined value at ./ast-rad-acc.pl line 293." It also show "Config file error" in the log file. Has any one meet this

[Asterisk-Users] TE410P and Fax Server

Hi, All Can I use a TE410P card to make a FAX server? Did anybody have some experiences to construct PBX and Fax Server in one box? Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com