search for: intertex

Hi All, I am attempting to setup Asterisk with sipcall.co.uk. They use Intertex kit to provide the SIP service. Unfortunately Asterisk cannot seem to authenticate against Intertex. Having provided SIP debug info the provider has informed me that Asterisk does not appear to support 'qop', 'nc' and 'cnonce' which are used to stop replay attacks. So, does...

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Hi, Has anyone got incoming IAX to work on the above router. I can call out, but incoming calls are not reaching the * box. Has anyone got this working? Could they give me some configuration hints. Thanks John

Hi All, Is anyone use the sipcall.co.uk 'professional' account with a UK geographic number? What do you think of the service? Alternatively, who else are you using to terminate a UK geographic number on asterisk? Thanks, Nathan. --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.529 / Virus Database: 324 - Release Date:

...do this for 6 phones.. I guess I would have to allocate a seperate port to each.. Also this could have major security implications.. I could use a STUN server but I have not found a free one yet.. I could use an ADSL router with a built in SIP proxy/registrar but the only one I have found is the Intertex IX66 and they seem quite expensive.. Anyone got any ideas or thoughts?? or even better experiences.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Besides ranch networks and borderware, what other SIP aware firewalls for the SOHO/medium market exists? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507) 6694-4780 ------------------------------------------------------------

Greetings list, Thanks to all who replied to my thread a few days ago "SIP devices with packet loss tolerance". One of the suggestions that came out of that thread was to replace routers at users' premises with ones that support QoS. I've used m0n0wall's QoS in the past with reasonable success, but it's quite a bulky and complex setup for deploying to remote sites which

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! I'm trying to set up the possibility for users to call my Asterisk from the net. The Asterisk is behind a Intertex IX66 in which a have set "Static domains" so it forward all calls for my hostname and external IP to the * box. when somenone calls at lars@<external-ip> it all works, but if they call lars@<hostnmame> it wont work. At my Asterisk console I get the following output; --<SN...

...eration not permitted. I get it on calls to the PSTN through my X100P (clone) as well as call connected through my IP telephony provider. I have also tried SJ-Phone and it happens with that as well. At the moment my asterisk server is on a public IP adress, and my client connects to it through an Intertex IX66 router. Before getting the router, I had dual NICs in the linux box and connected the client directly with the same problems... I've searched the lists but haven't been able to find any good answer, so any help is greatly appreciated :) /Anders -------------- next part -------------...

...t how would the ITSP end know where I am? I was hopeful that I could simply continue to have the register in sip.conf go directly to my ITSP and OpenSER would just "take care of it" and silently manage both sides of the registration similar to what happens on some commercial devices like InterTex's FireBird, but no luck. Can this be done? If so, what changes do I need to make to sip.conf etc.? If not, can you suggest an alternative so I can get the reinvites to work? In sip.conf I'll obviously need to make nat=no and canreinvite=yes. I don't think I need the following lines...

Hi All Has anything been done to fix the issue where the * box is sat behind a nat firewall? Regards Mark

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

...they attend SIPit and encourage them to participate - If you have collegues that work with SIP development, please forward this mail to them! Short facts: - Date: May 17-21 2010 (very beautiful season in Sweden!) - Location: Kista, Stockholm, Sweden - Host: Edvina and Tandberg - Sponsors: Ingate, Intertex, .se, Telio, Snom We are currently working to set the price and to be able open for registration. SIPit 26 has a Facebook event page at http://www.facebook.com/event.php?eid=340634688354 and a Twitter stream: http://twitter.com/sipit26 where you will get updates and be able to find links to the...

Hi, I've just (earlier today) updated from CVS so that I can apply the dtmf caller id patches. Unfortunately this has had an undesired effect. I have an intertex ix66 which up until the CVS update allowed me to register my * server with the ix66 for my local domain (eg sip.mydomain.com). Now it appears that asterisk gets totally confused and tries to register with itself! Anyone got any ideas? Thanks Andy 11 headers, 0 lines Reliably Transmitting: REG...

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

...tworks in sip.conf and those addresses will be excluded from the outbound proxy setting. This patch has been tested with a number of outbound proxies, but may still need some testing. It is really useful if you are using a SIP proxy as a NAT traversal SIP server, like the equipment from Ingate and Intertex. Please test this, and add your comments to the bug tracker! * SIP Outbound Proxy for Asterisk CVS head: http://bugs.digium.com/bug_view_page.php?bug_id=0002859 *** Read the daily news channel ------------------------------- There is a new news source for Asterisk news on the net, if you ha...

Should the following setup work? SIP UA---NAT---Internet---NAT---SIP UA If both UA's support STUN and report the external IP address in the SIP packet.. I am trying to get away from using canreinvite=no so that traffic can go directly between the UA's and not via the central server but I can't seem to get it to work.. Has anyone set this up and can give me some pointers??