similar to: asterisk with H.323 phone

Hi, I have one Q: why doesn't work h-extension in macro? I want to setup fax macro, to send tif-file after fax channel hang-up, but h-extension doesn't work in macro, so I must put them to global context. But in this setup, exten => h,1,system(/var/lib/asterisk/bin/mailfax ${FAXFILE} ${EMAILADDR} "${CALLERIDNUM} ${CALLERIDNAME}") mailfax binary will be executed after any

Hi, I have a bunch of NetPhones that I have bought from MeritCall some time ago for their service. How can I use this phones (supposed SIP phones) to integrate with a Asterisk Setup. I have seen a manual for a similar one but I don?t know If mine are hardcoded in some way. This devices are used by MeritCall , MamaKall, Vivophone etc. I found the manual for an exact same type here:

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

Hi all is there any way how can I simulate PSTN tone on asterisk. I

Hi all, I have one asterisk server with one ISDN BRI connection to PSTN, with h.323 support (oh323) I buy some voip phones, and I connect them to the same switch as asterisk server is; all is at the same TCP network. I need to route some extensions from my DDI (DID) line at asterisk to some voip phones, and also to do: outgoing calls from any voip phone to PSTN via asterisk, and intermediate

I've configured SIP softphone to work with asterisk n it's working fine..... but i m unable to connect my netphone IP phone.... I've connected my phone to LAN and assigned an IP address to it.... but how can i make call... plz tell me step wise..... _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today

I am using lme to handle repeated measurements. So far i can follow the examples from the book from pinheiro and bates. Now i get the problem , that i have "nested" repeated measuremnts, and i cant find out how to do the grouping part of the lme formula. 1.The simple problem ist that i have different Samples , from which i make repeated measurements (each sample is measured 6 times) and

Still looking for cheaper (under $250,-) alternative to cisco 7940 with features needed for corporate use, mainly: - shared phone book (e.g. via LDAP or XML browser in phone) - in-line power - missed/dialed/received numbers - integrated switch (voice VLAN support) I found only aastara/sayson phone (and Intracom/Netphone in the past), that has xml services anounced, but still not available, so

i read that asterisk supports iax,sip and h323 protocols.... i've used sip & iax softphones ... now i've a hardphone... an IP phone (Netphone) that supports h323 ..... i've compiled pwlib ,oh323 and asterisk -oh323 successfully ... but i m unable to place calls to/by my phone... i m confused whether to use h323.conf or oh323.conf and how ? i think it's different from iax.conf

Rich Adamson writes >>GMRS, FRS and MURS radios may not be interconnected with the PSTN (47 > > CFR 95.141). There has been a lot of talk from lobbyists to clarify this > > rule, but as it stands you could conceivably connect a *private* network > > to GMRS or MURS radios (you can't make any plugins or modifications to > > an FRS radio that isn't type

Hi! Useful advice, thanks! Actually I had been using foobar2000 to recode, because it just makes it so easy to convert multiple files while keeping the metadata (I confess, I'm a "tagger"). But it's easy to miss some encoder option when being presented some default suggestions in a dialog form... Apart form that I always had the impression that Opus could be quite smart

Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls "switching from simple_bridge technology to native_rtp" -- Executing [102 at mkg:1] Dial("SIP/101-00000017", "SIP/102") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-00000018 is ringing -- SIP/102-00000018 answered SIP/101-00000017

Dear R-Help, I'm using an ANOVA to determine if there is any variation that exists between my four different shark-types and their entrance behavior towards an apparatus. To do this, my coding was: df1=read.table("lemonentrance.txt", header=TRUE); df1 c(t(as.matrix(df1))) r = c(t(as.matrix(df1))) # response data r

Drive space for voice mail I've looked in the dimensioning information on voip-info.org but can't find any hard information on the amount of drive space the various codecs use. Since we would eventually like to support web-based voice mail retrieval, I'm thinking of the wav format. I've specced out 2x160GB drives in RAID-1 (software RAID via Linux) for the box. It will be

In the future could you please do some sort of visualization of your data, or at least provide the raw data in a machine-readable format so that others can do so? It is incredibly easy to come to incorrect conclusions when looking at lists of numbers because at any given moment you have a very localized view of the dataset and are prone to locally pattern-match and form a selection bias that

I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant

Jayson Vantuyl wrote: > Mmmmmm. I'm should patent that. > Something like "Patent for Using Spam > Filters to Block Stuff That Isn't Spam" or > something equally obvious. I like it. That won't make it even past the sloppiest examiner. You would have to wrap it up nicely and present it as something novel, like so ... ------------------------- Method and

Hello All, The attached patch allows openssh to specify which pam service name to authenticate users against by specifying the PAMServiceName attribute in the sshd_config file. Because the parameter can be included in the Match directive sections, it allows different authentication based on the Match directive. In our case, we use it to allow different levels of authentication based on the