similar to: call waiting from PSTN

I'm running into errors when using Enum lately. I can't figure out what the problem might be as I've had Enum up and running in the past. I'm running the latest CVS-Head compiled version. I've also tried using the new Enum function with the same results. When doing a lookup on a number that exists in the enum server I get the following results: -- Executing

All, I'm having trouble getting the X100P working. Lsmod shows : zaptel 179808 0 I did a . # modprobe zaptel and here is my zaptel.conf (comments omitted) __SNIP__ fxsks=1 loadzone = us defaultzone=us __SNIP__ Here is zapata.conf __SNIP__ [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no

Since e164.org added DNC and ADDRESS records my enum configuration has failed. Using both the old EnumLookup app and the new ENUMLOOKUP function, the lookups have consistantly failed since e164.org added E2U+ADDRESS and E2U+DNC records. Mar 6 17:39:44 WARNING[14222]: enum.c:235 parse_naptr: NAPTR Regex match failed. Mar 6 17:39:44 WARNING[14222]: enum.c:354 enum_callback: Failed to parse naptr

Does anyone have a better ENUM lookup handler than the built-in ENUMLOOKUP() function? The built-in function does not properly handle multiple return values such as: 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u" "E2U+SIP" "!^\\+1866(.*)$!sip:1866\\1 at tollfree.sip-happens.com!" . 8.9.9.3.2.8.8.6.6.8.1.e164.org has NAPTR record 200 10 "u"

Hello everyone, I playing around with ENUM and have configured * to query a few sources for testing purposes (fierymoon, e164.arpa, e164.org). I'd like to know if there is a way to query these servers manually (ie outside of asterisk via nslookup or equivalent) to find out if particular exchanges are listed with wildcards, so as to terminate calls to those prefixes (I'm not trying to

ENUM lookups at e164.org return a IP route for ALL toll-free numbers. I was surprised to observe that ALL toll-free numbers get a hit at e164.org. It appears that ALL toll-free prefixes have been delegated, thereby publishing an IP route for YOUR TOLL-FREE NUMBERS, my toll-free numbers, and even toll-free numbers that have not been allocated. :-) See below Should I care? Even though this

This has been giving me a headache for a little while, so I thought I would throw the question out to the list. Imagine there is a database lookup function that takes place that, given a number, returns call routing information. It might be a postgresql lookup, it might be an enum lookup. If I have something like: exten => _.,1,LookUp(${EXTEN}) exten => _.,2,Dial(${RESULT}) Then the

Ok, so you've read the Wiki and gotten call routing using ENUM to work (http://www.voip-info.org/tiki-index.php?page=Asterisk%20E164%20Call%20Routing) with your own ENUM-alike domain, e164.example.com. But how do you populate it with data? You can do it manually, but that gets very tedious very quickly. Or you can use the nifty DDNS updating program that comes with bind9. The first thing is

hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian --------- gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify: * srvlookup=yes|no * pedantic * canreinvite=update|yes --"update" seems new Being curious, especially for "srvlookup" functionality... /O

Hi, I'd like to setup little private enum server. Any more info on how to do that ? Regards, Rob.

I am having trouble with chan_zap.so not loading. When I load it from modules.conf, Asterisk bails out without any error message. When I load it from the console, it just says "Unable to load module chan_zap.so" no matter what verbose level I am using. dmesg says: Zaptel Version: 1.4.4 Zaptel Echo Canceller: MG2 Freshmaker version: 73 Freshmaker passed register test Module 0:

I'm trying to implement direct_media between multiple peers and an uplink provider, all of whom have direct_media=yes configures. For originating calls to the uplink provider direct_media=yes works like expected. SIP flows through asterisk, rtp doesn't SIP: enduser <-> SBC <-> asterisk 13 <-> uplink RTP: enduser <-> SBC <-----------------> uplink SBC

I'm trying to configure sip2sip, which says: http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk "Asterisk, is currently unable to handle more that one result for a DNS SRV lookup, and the Asterisk configuration needed for getting it work with the SIP2SIP service is not trivial" It then gives a complex multi-section workaround in SIP. I remember reading there'd be

CentOS-6.5 (FreePBX-2.6) Asterisk-11.14.2 (FreePBX) snom870-SIP 8.7.3.25.5 I am having a very difficult time attempting to get TLS and SRTP working with Asterisk and anything else. At the moment I am trying to get TLS functioning with our Snom870 desk-sets. And I am not having much luck. Since this is an extraordinarily (to me) Byzantine environemnt I am going to ask if any of you have gotten

All, If I have video phones behind an asterisk server (with 2 network cards) and all the phones have extensions. Internally everything works great. Now for people that want to call my video phones external to my office is there a way to do that? On the extenal persons phone enter an IP/EXTEN where IP is my server and not the phone? Can that work? Would I have to have PUBLIC IP address for every

Friends, The following mail was sent earlier to asterisk-dev and did not cause the amount of discussion I hoped it would. Now that we have a way to secure signalling in IAX2 and SIP in Asterisk svn trunk, we need to start working on the concept of a "secure call" - or does it really matter? In SIP, there's a specification for how I as a domain owner can request all calls to

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1&t=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate it all here, I've posted my logs and conf files on that thread, too. Problem is that while

I've been tinkering with ENUM and found that the lack of a debug message in enum.c that says it has actually succeeded in resolving an address is a bit of a nuisance. It makes it difficult to see if failures with ENUM are due to problems with parsing NAPTR records (in enum.c) or mistakes in extensions.conf An extra line of debug information would be much appreciated! Iain