similar to: Back to back E1 - asterisk <=> toshiba pbx - Call droping issue

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

Hi All, Thanks for every one who helped me on this regard. I think i was able to rictify the problem. what i did is remove callprogress=yes usecallinpres=yes and restart asterisk. Today i didn't report any drop calls. Many thanks for Eric. :) I hope this situation will continue. Regards, Vidura. On 3/8/07, Vidura Senadeera <vidurased@gmail.com> wrote: > > Hi, > >

Hi steve and All, I'm attaching cat proc/interrupts out put, lspci -bv out put, zapa.conf, zaptel.conf for your information Thanks so much for the feedback and I do accordingly. Hope to get rid off this isue any how. To day also reported 10 call drops within 2 hours of period. fook forward to have your support on this regard. Thanks & Regards, Vidura Senadeera, Network Engineer,

As these problems are very time sensitive and frustrating, I suggest you document each change you make and do them one at a time so you can actually know what the problem was and not introduce new problems in the process. Find someone who is on the phone quite a bit and will give you an honest evaluation of the call dropping situation (unless you yourself are experiencing this issue too).

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk <-> Digium TE110P <-> E1 card in toshiba pbx <-> Toshiba PBX* A =============================================> B C <============================================ D Asterisk PBX and strata PBX connected using back to back E1 cross cable. Physicall connectivity

Hi All, I would like to explain the layout that i am trying to achive. I am so helpless on this regard. So here is the story ........ " This is with regard to the setup which you can find at the "Asterisk The Future of Telephony" , chapter 11, page # 196-197, I am attaching the picture for your information. Now I am taking a challenging step to of integrate IP PBX with our

Dear All, I would like to get community's feedback with regard to RAID1 ( Software or Hardware) implementations with asterisk. This is my setup Motherboard with SATA RAID1 support CENT OS 4.4 Asterisk 1.2.19 Libpri/zaptel latest release 2.8 Ghz Intel processor 2 80 GB SATA Hard disks 256 MB RAM digium PRI/E1 card Following are the concerns I am having I'm planing to put this asterisk

Hellow, * I have a problem with dial up signalling. currently I have configured asterisk server and E1 card to ISP. then other side I am having ATA to PC for connecting internet through DialUP connection. is it possible and please send me the procedure how I can do it ?? * ISP< <-> Asterisk <-> ATA <-> DIALUP -- Thanks & Regards, Vidura Senadeera, Sri Lanka.

Dear All, I'm integrating avaya commuication manager difinity ver 1.0 with asterisk using B2B E1. following are the details of my H/W, zaptel configs and software installed. Digium TE110p asterisk 1.2.19 cent OS 4.4 zaptel 1.2.18 libpri 1.2.4 etc/zaptel.conf span=1,0,0,cas,hdb3 bchan=1-15,17-31 dchan=16 when i ztcfg -vvv im having this error message and the E1 is not getting up. "cas

> > > Hi, > > > I checked by changing to from-zaptel, but no luck yet. Pls guide me on > this. > > Regards, > vudura senadeera > > > ------------------------------ > > > > Message: 9 > > Date: Fri, 19 Jan 2007 16:47:18 -0000 > > From: "Robert Jenkins" < raj@jrw.co.uk> > > Subject: RE: [asterisk-users]

Dear All, I am experiance a issue with my IAX clients. I have upgradeed Asterisk to 1.4.28 After then IAX clients are not working and It's not registering even. Please help. Asterisk previous version - 1.4.26.1 ( for this worked fine) FreePBX version - freepbx-2.5.2 -- Thanks & Regards, Vidura Senadeera, Sri Lanka. msn/yahoo/skype Ids - vidurased -------------- next part

> > Hi, You can achieve this by integrate CCM and asterisk using SIP trunk. In CCM you can create SIP trunk, After creating SIP trunk in between CCM and asterisk, you have to configure dialplan on CCM to pass the calls to asterisk. One the caller id comes to Asterisk you have to use extension.conf to route the calls. You can also try with freepbx GUI to configure inbound route, it makes

> Hi Gareth Blades & Doug, > > Thanks so much for for the feedback. I have searched on lot of documents > but couldn't able to find clear answer regarding it. > > I hope you guys replies are very much help all in aterisk community. > > > Thanks & Regards, > > Vidura Senadeera, > > Network Engineer, > > Debug Solutions > > Sri Lanka .

Hi Carlos/All, Thanks for your reply. I can remove dchan=16 from zaptel.conf But according to the documentation of Digium and sangoma they mentioning to use dchan=16. Are there any specific reason you have experiance regarding this and I am confusing that what this is included to the documentations. Regards, Vidura. On Wed, 2007-09-05 at 20:26 +0530, Vidura Senadeera wrote: > Dear All,

Hi! Just ashort question - obviously I am too stupid too find the answer on the net. :-) I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have to do? Just install it over the existing version? Do I need to backup the configuration? Will I need to reconfigure the source or will the new version "import" my old settings? Will I need to update Zaptel and Libpri too?

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Hi, I am currently integrating our company's Toshiba PBX with the Asterisk version 1.2.1. I bought Quad T1 card, and making the port 1 to connect to PSTN PRI (use "pri_cpe" in zaptel.conf") and making the port 3 to connect to Toshiba PBX (using "pri_net" in zaptel.conf). The first stage goal is to just adding the Asterisk relay between PSTN and Toshiba system. The

Hi all, I'm dealing with a Toshiba PBX - DK 40 I think it is...The problem is this: If someone dialing in from the PBX (on one of the two X100P cards) goes into a MeetMe conference...Everything is fine, until they hang up - especially if there's music on hold - the line is held open because Asterisk does not know the far end hung up. I have it set to fxo_ls right now for signalling.

Is there anything special that anyone here has had to do to get an Aastra phone (on the Internet) to talk to Asterisk behind a PIX firewall? Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk server. The Asterisk server's RTP.CONF is set to use 10000-20000. The phone registers, and will place AND receive calls, however, no audio is passed. The phone is an Aastra

Hello everyone. I''m using drag and drop on my site, I''v notice strange effect about droping. When I hold element over the droppable area, and drop on it, i see that element not exacly drop on the droppable area, I see something like element coming from outside of the screen, or coming from diffrent position, it''s look like element slide and then is dropped on the