similar to: TDM10M && Siemens Euroset 2015

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Please help find the cause of strange behavior res_pjsip. Making outgoint call to other sip server (CommuniGatePro), my asterisk suddenly sends BYE after picking up! Partial log of an outgoing call with full debug is attached and on web: http://pastebin.com/tLNCpx4d No diagnostic messages why asterisk suddenly decided to hangup i don't found :( There are suggestions or strong belief

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

Hi, I've noticed every time someone does an rsync-request on my ftp-site (which also provides rsync as mirror method), rsyncd creates a filelist. This is a quite IO and CPU intensive procedure, especially for things mirrors like FreeBSD with lots of little files. I was wondering... is there a way to cache that filelist? Our mirrors are updated once, or twice a day, it could speed up

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

MixMonitor seems to work: -- User hit '*3' to record call. filename: auto-1250792853-24-22 == Begin MixMonitor Recording SIP/snom2-084c4ec8 /var/spool/asterisk/monitor/auto-1250792853-24-22.wav exists now. Recording a call without mixing fails. > User hit '*1' to record call. filename: wav,auto-1250793354-24-22,m TOUCH_MONITOR_OUTPUT is set to

Hi, we have a stange problem here since samba 4.12. Users on specific clients are not allowed to access a share after reboot of the client and in very rare cases suddenly during use. When we restart the smb service after the access denied message on the client for a share, the user is allowed to access the share. So there seems no problem with permissions. Are there any known reasons for

Hi, I'm practising to do a classicupgrade of my samba3 domain. When I run the upgrade, I get this error : unpack_nt_owners: owner sid mapped to uid 0 idmapping sid_to_xid failed for id[0]=S-1-5-32-544: NT_STATUS_NONE_MAPPED unpack_nt_owners: unable to validate group sid. set_nt_acl_no_snum: fset_nt_acl returned NT_STATUS_INVALID_OWNER. ERROR(runtime): uncaught exception - (-1073741734,

as u have said in a reply mail below could you please send me the sample config files.. please as i was trying to do LDAP for BDC/PDC for quite sometime and un-successfull also. Your below gives me very good thought trying it agian. I also hope ur sample config files might come great help Thanks in advance with best Regards Yeri samba-request@lists.samba.org wrote: >Send samba mailing

Hello guys! I hope you bare with yet another newbie on the list! :-) I am trying to setup an asterisk installation in between a Siemens HiPath 3800 and my local carrier (Telefonica/Brazil). Both running R2. ISDN is not an option on the carrier. :-( I could apparently setup both E1s just fine according to zttool (both OK with no alarms) but, and this is where it starts to get

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com> > Kevin P. Fleming wrote: > > Olivier wrote: > > > > > >> 2. R Hook-flash key is now available to transfer calls. > >> In s450IP web management server, its defaults settings are : > >> Application-type: dtmf-relay > >> Application-signal: 16 > >> >

I recently bought an edge modem and I need help on how to install it on CentOS 5. I tried working my way round it by installing windows as a virtual machine and then installing the modem but it kept on telling me that the USB device has malfunctioned. When I install on Windows XP based computer (no VM) it works fine. Any help out there? -------------- next part -------------- An HTML attachment

I tried a work around i.e. installing Win XP as guest using VirtualBox and then running the modem from win XP seamlessly but ran into another brickwall. it gives the following error when I try to enable the Samba 75 for win XP: Failed to attach the USB device Siemens AG Wireless modules to the Virtual Machine Windows XP. Failed to create a proxy device for the USB device (error VERR_READ_ERROR).

Good day all, we have following setup: Debian Etch 64, Asterisk SVN-branch-1.4-r66244 with mISDN 1.1.3 and 2 Digium cards B410P. Our customers calls in the office through ISDN lines and then get a possibility to join meetme conference. It works well except when customers are using SIEMENS phones (not IP): DTMF is not recognized. Does someone have an idea on what could be the problem with

Hi, I will like to know if anyone would guide me about how I can to interconnect one SIEMENS HiPATH 3700 with Asterisk. HiPATH have VoIP card and my idea is to do one un IP trunk between them so we would to transfer calls and services (voicemail, IVR,..) between both. We havent PRI ports unused in HiPATH so cheapest method of interconnection is one IP trunk. Any help or comment about will be

Hi all, I want to use asteriks with my siemens hipath 1120 hardware. Is it possible? Thanks -- /** * @AUTHOR At?f CEYLAN * Software Developer & System Admin * http://www.atifceylan.com */ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101125/a5b9e9f6/attachment.htm

Hy all of you, Is anybody has a tutorial for integrate a siemens gigaset as180 and connect it to Asterisk. I've searched a lot and didn't found something concluding. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110611/17522d7a/attachment.htm>

hi all, anybody have a siemens hipath 3500 with a sm2/pri card? because i need to connect to my box TE110P (e1) and i dont know how is the mode in the pbx to change it. thanks -- .- Pablo Allietti LACNIC

Hello, My goal is to provision C450IP or S450IP models. Has anyone a hint to provision them from configuration files ? Usually, we use dedicated menu embedded inside Gigaset handset. An http server also exists but I couldn't find any dhcp-tftp combination to configure them. Any clue ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: