similar to: zaptel/dahdi settings for singtel E1 line

Trying to install wanpipe 3.5.18. No errors during compile. But when I reach the point where wanpipe and dahdi_cfg is started, I encountered an error. Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid

http://www.smh.com.au/articles/2004/04/05/1081017104255.html SingTel ready to break into web telephony April 6, 2004 Singapore Telecommunications is teaming up with US internet phone start-up SIPphone to offer low cost, and in some cases free, phone services over the web. The deal, expected to be announced today, will allow SIPphone - started by MP3.com founder

Hi, Has anyone successfully configured a HFC ISDN card with Singtel's ISDN-2 service? If so, can you share the settings required? Thanks in advance. KokMeng.

I am currently testing a 57i unit. No problems configuring the phone's config via phone/web UI. We are trying to avoid using the web UI, the reason is it will take a long time typing the softkey xml applications URIs on each phone, so we chose TFTP. Tried configuring the phone via a TFTP config server, but no changes took effect. I wonder why it doesn't work with TFTP even if I was able

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now more than 70% of call get good voice but some of calls having issue for callquality and other voice related issues. now my question is that is there any

We're encountering no audio issues in ConfBridge. Only the moderator and the 1st invited have audio. When the 2nd invited number picks up the phone, only the announcement is heard, then followed by silence. This occurs frequently. Only 1 out of 5 test calls do not have this problem. We were hoping that switching from MeetMe to ConfBridge will solve the no audio issue, since there's no

We have set the do not disturb feature on our polycom phone such that incoming calls will not be rejected and sent to 'Busy' status. The user can still toggle dnd on/off, but incoming calls will still get in, indicated by the blinking light and the screen status. We were able to do that by setting call.rejectBusyOnDnd="0" in sip.cfg. However, the problem is that there is still

Hi, I have just installed Slackware64 13.0 Linux as a HVM guest with Xen 3.5-unstable changeset 20143 pv-ops dom0 kernel 2.6.31.4. For more details, please refer to the following link: http://enmingteo.wordpress.com/2009/11/01/picture-tutorial-how-to-setup-slackware64-13-0-hvm-domu-with-xen-3-5-unstable-pv-ops-dom0-kernel-2-6-31-4-in-fedora-11-x86-64-dom0/ There are 83 screenshots so far at

Hi, I have just installed Slackware64 13.0 Linux as a HVM guest with Xen 3.5-unstable changeset 20143 pv-ops dom0 kernel 2.6.31.4. For more details, please refer to the following link: http://enmingteo.wordpress.com/2009/11/01/picture-tutorial-how-to-setup-slackware64-13-0-hvm-domu-with-xen-3-5-unstable-pv-ops-dom0-kernel-2-6-31-4-in-fedora-11-x86-64-dom0/ There are 83 screenshots so far at

Hi, our Asterisk is connected to an E1 port. So we are using the DAHDI-Driver. Please , how do I tell the driver/Asterisk to wait for overlap digits for in-calls? I found the option "overlapdial=yes" but I did not try yet. Is that "my" option? Is there any option for setting an timeout? Thorsten

hi, i was able to get this working with google talk. i entered myusername@gmail.com using the gtalk2voip.com website's "invite" box, and as a result, saw a request from service@gtalk2voip.com to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten => 3501, 1,

Has anyone managed to get gtalktovoip working at all? If so please explain. http://www.gtalk2voip.com/faq.shtml 2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ? A: This is a major feature of our gateway and it is very easy. o GTalk: user@domain.com can be reached by calling to sip:user_at_domain.com@gtalk.gtalk2voip.com o MSN: user@domain.com can be

Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I" from voismart (http://www.voismart.it/) but the driver is very bad (compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable and have an echo cancelltaion feature. And of course it

Hi List, Were upgrading our network switches and need to create multiple VLAN groups, but since our Squid Proxy (Transparent Proxy) Server should be accessible to all VLAN groups we need to setup a trunk grouping inside our Squid Proxy Box. Is anyone has a documentation or code on how to implement trunk grouping? Your thoughts will be highly appreciated. Regards, Malvin

Hi,all I configured two Asterisk PBXs: 1.4.X and 1.6.X. All relevant ports are opened, externip is configured in sip.conf. I think, that all relevant configurations are checked. But, no voice hear between local and remote extension. What I need to check, configure in router and PBX for resolving this issue ? How I can to install and configure STUN server ? Thanks, Oleg . -------------- next part

Hey Guy, I have quick question. I am purchasing Sangoma A102D card but i am confused between PCI and PCI Express. Which card would be good for me. Definitely PCI Express is advance but i just want to know is there any major difference, like quality, performance etc.. -Satish -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear All, I have created a virtual high performance computing (HPC) cluster of 6 compute nodes with MPICH2 using Xen-based Fedora 11 Linux 64-bit paravirtualized (PV) domU guests. Dom0 is Fedora 11 Linux 64-bit. My Intel Desktop Board DQ45CB has a single onboard Gigabit LAN network adapter. I am able to bring up the ring of mpd on the set of 6 compute nodes. However, I am consistently

Dear All, I have created a virtual high performance computing (HPC) cluster of 6 compute nodes with MPICH2 using Xen-based Fedora 11 Linux 64-bit paravirtualized (PV) domU guests. Dom0 is Fedora 11 Linux 64-bit. My Intel Desktop Board DQ45CB has a single onboard Gigabit LAN network adapter. I am able to bring up the ring of mpd on the set of 6 compute nodes. However, I am consistently