similar to: Asterisk with iptel.org

Hi all, I have to install 25 IP Phone in some building. I want just basic IP Phones like: Cisco-Linksys SPA922 u$s 146 Grandstream GXP-2000 u$s 105 Snom 300 u$s 119 The most valuables parameters for me are (in importance order from high to low): - Stability (device don't hang in any way) - Voice quality using G729 - Provisioning So what device do

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) --

Hi, how I can connect Astrisk to my iptel account??? I have try to this configuration, but it doesn't work: In sip.conf: register => my_account_name:xxxx@iptel.org [iptel.org] type=friend host=iptel.org fromuser=my_account_name secret=xxxx nat=yes in extensions.conf: [fromiptel] exten => my_iptel_number,1,Dial(SIP/phone1,20,r) [toiptel] exten =>

Hi all, I'm using Asterisk 1.2.5 and , for some reason, when I install it, the module app_meetme.so didn't install. Is there some way to download that module, and add it to asterisk without re-install it? Thanks in advance Sebastian

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

Bug in the register= code; see details below from the developer of "ser" (SIP Express Router) Apparently, ACKs don't need to be sent on OK's to REGISTERs. Plus, malformed data somewhere... no details on that, though. JT >Date: Sun, 23 Feb 2003 23:54:07 +0100 >To: John Todd <jtodd at loligo.com> >From: Jiri Kuthan <jiri at iptel.org> >Subject: Re:

I'll admit it. I'm a asterisk newbie (but no stranger to telephony). The setup is simple: two Grandstream BudgeTel 100 phones (SIPPhone specials) on a private segment calling to a Linux box acting as the segment's firewall with a leg on our public network. The phones are setup as SIP/phone1 (x1000) and SIP/phone2 (x1001), respectively (thanks to the Asterisk HOWTO). Getting IAX

Hi all, I'm new in the forum, and although I have some experience in Asterisk, I've never work with Asterisk FXO, FXS, E1... cards. I have several costumers with ATAs working with my SER. However one of them bought an Alcatel PBX OmniPCX Enterprise and now he wants I give him a E1 interface for interconection with its new PBX. I understand I need a E1-IP gateway which could be Asterisk

I can not register via sip to iptel or sipgate. i do not unterstand why.. but i am new to asterisk. Iam behind a susefirewall2 but asterisk even do not register if it shut down. No answer seems coming back. thx for help. nico here is my config if anybody can help: ----------------------------------------- [general] port = 5060?????????????????????; Port to bind to bindaddr =

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Hi all, I have around 100 SPA2100 registered in my provider openSER. I'd like to add an Asterisk registered into openSER, to the network, to deploy voicemail service for those SPAs. Due to administration access levels, I have no access to SER box, so I'm wondering if that possible: - Some foreign user (say A) calls one of my SPA (say B). - B don't answer. So.. B SPA is setted up to

I can not register via sip to iptel or sipgate and do not see sip into ethereal. I do not unterstand why thats Wudu .. but i am new to asterisk and sip. I am behind a susefirewall2 but asterisk even do not register if it is down. The asterisk is running onto the machine witch is connected to the internet. No answer seems coming back from iptel (sip debug in asterisk). Ports are open (5060,

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten =>

I am trying to make an outgoing call using an iptel account using Asterisk. I have followed a how-to for asterisk and iptel found at http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER I am getting the following error message: Got SIP response 403 "That is ugly -- use From=id next time (OB)" back from 195.37.77.101 I'm not quite sure what that means. Does

Do some reading about contexts in *. Basically, you want all "public" sip requests to land in a dialplan context that has no access to PSTN, and requests from your own SER box(es) to land in another context (that DOES have access to PSTN). You can achieve this by adding an entry to your sip.conf for your SER box with it's IP address (and context) specified. ----- Original

It's been down the last 5 hours at least. Anyone know what the problem is, or when it will be back up?

Hello. I have two records in dialplan: exten => testA,1,Dial(PJSIP/outgoing/sip:thetestcall at sip.linphone.org) exten => testB,1,Dial(PJSIP/outgoing/sip:thetestcall at iptel.org) Calling testA works fine while testB fails with "CONGESTION". Adding debug for console shows that pjsip_resolver.c does `New queries added, performing parallel resolution again` for linphone after

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

Hello Group, I am new to asterisk but learn a lot about it to this mailing list and wiki currently i am facing problem about sip phone i have "PA 1688" chipset ip-phone and i have iptel.org sip account i registered locally and through iptel.org comfortably my problem is that when i called from my sip phone to analog or any number after connection my sip phone generates very load scartchy

Hello everyone! I still am not much further along with my sip calling. I changed my sip.conf taking suggestions from the net (voip-info.org in particular). I changed iptel's position from friend to peer. I turned on and off nat, I chose different codecs in first place, entered my outward IP as fromdomain and uncommented the register directive with correct values. All I get is two