similar to: Fw: Fw: Questions and finding

----- Original Message ----- From: "Jess Magnaye" <jess@arretni.com> To: <wipe_out@users.sourceforge.net> Sent: Tuesday, January 06, 2004 3:19 PM Subject: Re: [Asterisk-Users] Pls confirm > Is the format "allow=g723.1" in sip.conf valid? > > somehow i cannot get it working to do g723 passthru. also, i've read that > doing g723 will disable

I installed * to primarily test its voicemail feature. I installed it on a server WITHOUT any telco board (i.e., digium). Installation looks ok, however I am having problems. MY SETUP: 2xATAs are configured to use * as GkorProxy Asterisk is registered to my SER SIP/RTP Proxy 1.) First test - ATA1 calls to ATA2. When voicemail starts playing, it just cuts-off after 5-10seconds (consistently).

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

On Thursday, January 22, 2004 9:55 PM, Jess Magnaye [SMTP:jess@arretni.com] wrote: > in telco world, there's like 64kbps per channel and voice can be > carried on a 16kbps channel. is it possible to configure asterisk to > make 4 extensions (ATAs example), to call out using single FXO port > at the same time? if that is possible, then is it also possible to > make t1-pri to

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

Hi! Yes you're right (for windows), but I found this thread http://www.mail-archive.com/asterisk-users@lists.digium.com/msg10695.html and that works! The first card is connected to a normal Telekom NTBA, the second to an internal PBX. There have to be a possibility to configure multiple ISDN cards (e.g. AVM B1 PCI) through capi.conf. How? Or does chan_capi support only one ISDN-Card?

Can someone on the list confirm if Asterisk can do g723 or g729? when connecting to provider? or it is only supporting g711? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040106/1d6c78cb/attachment.htm

supposed to be able to give you much help with such little info anyway), I can only guess that since you are using the 's' extension, you are in a macro ? If so, try scrolling down the wiki page to the example using '[macro-inbound]'.<br> <br> Rob<br> <br> Jonas Kellens wrote: <blockquote cite="mid:4C17C4A1.8020404 at telenet.be"

I have looked at several IAX and SIP soft phones but I have been disappointed with the sound quality on my Windows XP Pro PC. Also the GrandStream problem is that they don't yet support headsets. When I turn auto answer on and I dial in it instantly picks up with the speaker phone. But if I have the handset picked up when a call is coming in the line is busy. That means that the phone itself

Funny - I downloaded the latest Asterisk CVS, and it's pretty much working. Will report when I have some more success. PaulH -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Tuesday, 28 September 2004 9:46 PM To: Paul Hales Subject: Re: [Asterisk-Users] Leader IP10S Hi! > I have been lent a Leader IP10S phone (SIP) for

I am just wondering if this is normal. I have my * running for a week now and I'm still testing its interoperability with other voip provider (in sip using codecs other than g711). yesterday, i changed my linux's (RH9). and since the new ip i assigned is located on a different site, i have to shut it down and move it physically. after that, i cannot run my * anymore. i am getting this

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not

Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Am I doing

Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment

I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the "host=" IP, instead of the "outboundproxy=" IP as it should, verified with tcpdump. I've done a

Anybody on the list who implemented Cisco ATA + * + Cisco 2600? I cannot get my calls from ATA to terminate to the Cisco gateway via *. I am not sure if it is my hardware problem. I'm getting the following "codec negotiation problem" from Cisco. 23:39:08: Unexpected VoIPCodec Type :g729br8 23:39:08: Unexpected VoIPCodec Type :gsmefr I appreciate any help I can get. Thanks.