similar to: Zap timeout not occurring

Hi, I am using real-time, oh-0.7.2, G729 Calling from (SIP)UA through asterisk towards h323 devices or the other way round, I get only one-way audio. Called party can only talk, caller can only listen. Calling SIP to SIP is ok. All devices are on official IP addresses. (no NAT) Regards, Shaoul Jacobson Senior VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail

Well here's a suggestion - a little crazy - but works... Most equipment is taking the 120vac and converting it into DC voltage. So why not just feed it DC voltage directly??? We had a situation where our field techs needed to test dsl circuits and voip ata from the demarcation point outside a house or business. A UPS might have worked - but the down conversion of 12v dc battery in ups up to

Hi, cisco's phones are VoIP only polycom build (video-) conferencing devices. One Cisco model (7930 I thnk) is a polycom in disguise. The code is not 'cisco-like' (at least the version I had. Both brands make very good quality equipments. Good sound, good support, ... Regards, Shaoul Jacobson VoIP Consultant Tellink Tel : +32 3 201 96 36 Fax : +32 3 227 09 81 e-mail

Hi, pwlib 1.6.6 & downloaded & ./configure & make it as written The same with openh323-1.13.5 Downloaded & patched make & ./configure & make it as written Then with asterisk-oh323-0.7.1 Downloaded (I used u file there to patch openh323) Made some changes in the Makefile to adjust directories Then 'make' I got an error in chan_oh323 : asterisk/channel_pvt.h

Hi, I have the same problem with cvs head. (1.0.6) See http://www.inaccessnetworks.com/projects/asterisk-oh323 And https://skylab.inaccessnetworks.com/mantis/view_all_bug_page.php (issue 00...008) some 'patch' files are included. I am a newbie to linux and asterisk. I do not want to blow my config. Please give me a feed-back if those files helped you and how. Also if you have a

We have one Swissvoice IP10S running SIP firmware. Recently, I've been getting these messages: Mar 1 13:59:44 NOTICE[20933]: frame.c:128 ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end Any clues off the bat? I'm still researching other stuff.. Thanks, Matthew

Hi list! Sorry for this slightly off-topic message but does anybody know if the standard for ISDN BRI is the same in Spain as it is in the rest of Europe (or the Netherlands). Will a standard HFC-S card work?

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

Dear Fellow *-ers, First, you guys are fantastic. Keep fighting the good fight. Second, it sounds like comments in the code are coming, which sounds welcome by all, even those of us who couldn't code their way out of a papersack, but who need to read the source. Last, I might be traveling to Europe (from US) & want to tow along hardware & haven't done this before & was

I have the following configuration and am obviously missing something small that is causing * not to work as expected. I have the following defined in sip.conf [ccme-in] type=peer host=10.0.9.1 context=devel_in disallow=all allow=alaw nat=no canreinvite=yes qualify=yes and [devel_in] is defined in extentions.conf However when I try to call via the dial peer I have configured on the cisco

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Has anyone been seeing IO lockup problems on EL4? I've tried multiple IO scheduler options (elevator=) in the boot... I'm seeing the same behavior regardless. Independent of hardware. Whitebox ATA, HA enclosure with dedicated SCSI, megaraid RAID hardware, Dell 2850s... same behavior: A semi-busy system will suddenly go into some kind of IO la-la land where nothing can be written

I have just started developing asterisk, and am trying to start simple. I have a X100P device and an S100U device. I am trying to use the examples provided, where I add a few lines to the /etc/zaptel.conf, /etc/asterisk/zapata.conf, and /etc/asterisk/extensions.conf so that I may connect an analog line to the X100P and an analog phone to the S100U. When I dial the analog line, it should ring

Greetings, I am porting the openssh-portable 6.8 release to the HP NonStop (NSE) platform. Prior versions were no real problem, with minor tweeks. However, with the inclusion of regress/netcat.c, which depends on arpa/telnet.h, we have an issue. Unfortunately, the platform does not have this file, nor anything like it - telnet is done rather differently. We do have a version of netcat (0.7.1

Hi all, I have a voice menu system ( Outlined below ), and I'd like to give the user some feedback when they dial an extension ( ringing, music, SOMETHING ). As it stands, when a user enters an extension from the menu system, they hear silence while the line rings. I even tried including the Ringing application before calling my macro to dial the phones, with no luck. Any help is

Would it be considered normal that a system behind a shorewall box that was setup for proxyarp and able to be reached from the trusted side of the net just fine on the proxyapr ip address would if it were to talk out to the world show as traffic not from the proxyarp address but the firewall''s own address or the masquerading ip used by other zones? We had not really noticed this as an

I'm not sure if this is the appropriate place for this suggestion but I'd like to get some discussion on the subject. I feel that AppDB does not provide enough information about the particulars of the system and user to be of a lot of value in deciding what works, what doesn't and why. I'd like to suggest that AppDB capture some additional information: 1. User experience: e.g.

Hi everybody In advance sorry for my bad english and if my problem was already exposed (I didn't find any tips in the mailing list archive. Bad luck) I have some questions about asterisk 1.6 release : 1) how can I do a n+101 priority jumping if a SIP canal is busy ? I read that the general parameter "priorityjumping" is depreciated in the 1.6 release and I already try the

Hello everyone I have configured samba4 as per the details prived on samba how to homepage. Now i can successfully add my windows XP and windows 7 machine to the domain. I logged in to windows XP machine as domain administrator and created a user using dsa.msc The user is able to login on windows XP machine but when I tried to login on windows 7 this is the error which I get The trust

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]