similar to: a few questions about sip implementation

Mark has fixed the REGISTER issues to be more RFC compliant. I've created a new thread so that those of you who got bored with the old thread might read this new one. The feature that has just been added was added a while ago, but now it actually seems to _work_. :-) If you have a SIP server to which you are trying to REGISTER, and they demand valid domain (the part after the

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming

I have a SIP trunk - calls going out work fine. Trying to setup an incoming call with a DNIS When I dial the number - I see nothing on the CLI. The person says the server is returning 401 How do I debug that. Using asterisk 18.8.0 Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm getting the below error when trying to compile the latest git. gcc -c -I. -I. -I../../include -I../../include -D__WINESRC__ -D_REENTRANT -fPIC -Wall -pipe -fno-strict-aliasing -Wdeclaration-after-statement -Wstrict-prototypes -Wtype-limits -Wwrite-strings -Wpointer-arith -g -O2 -o registrar.o registrar.c registrar.c: In function 'DllGetClassObject': registrar.c:747: error:

Hi, I've been building wine from source for years, originally because I needed some custom patches, nowadays out of habit. Just now I'm trying to build 1.3.8, but I get an error during build: make[1]: Entering directory `/home/peter/src/wine/dlls/atl' gcc -m32 -c -I. -I. -I../../include -I../../include -D__WINESRC__ -D_REENTRANT -fPIC -Wall -pipe -fno-strict-aliasing

Unable to determine IP address from host name "www.hdt-project.org" Getting this today? Not sure what issue is? I paid for the renewal back in 08/04/2016 and and in 2015, so the domain should be current? But the whois seems to show it is expired? Went to the gandi site, and it doesn't show a renewal option or anything? whois hdt-project.org [Querying

Hi, I'm currently trying to connect an Aastra 57i to our Asterisk Server. The strange thing is, that altough I have definitely entered the correct IP address of the server, the phone doesn't even attempt to register. Here is the configuration file (local.cfg) of the phone: firmware md5: dee6e938b469e217a87138076f47fe41 boot count: 1 tone set: Germany language 1: German time server1:

Hello, I have recently switched from having a dynamic IP address and using a DNS service like zoneedit and dyndns to having a static IP address. How do I stop having to use these DNS services and use my own? I tried changing the DNS servers at my registrar but it won't accept my server. TIA

try "sipsak -M -O desktop -B "foo" -s sip:<user>@<registrar> -H <ip of registrar>" the trick is to specify the "-O desktop" parameter + the "-H <ip of registrar>" parameter. Sipsak fakes the host-header of the registrar so that the Snom thinks it is coming from your Asterisk server, then lets the message through to the

Hello All, Very sorry about WAY off-topic query, but you folks really are one of my most International subscribed groups. I am looking for a recommendation for a domain name registrar I can register my .co.za domain name with that won't 'yank my chains'. I tried a couple attempts at registering and found some hidden fees along with the insistence that I had to host my DNS with

Hi! Then I phone other phones with my Zyxel 2000W (WI-FI) it just hang up when I answer the phone I am ringing. It works fine if I call the 2000W from other phones. I have tried many sip settings. I use this now: [205] type=friend username=205 secret=passwd205 callerid="Zyxel" <205> host=dynamic context=local nat=yes canreinvite=no disallow=all allow=g729

hi, Is asterisk a registrar server. thanks, satish Confidentiality Notice The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, please notify the sender at Wipro or Mailadmin@wipro.com immediately and

On 2/12/19 10:55 PM, Alice Wonder wrote: > DNSSEC keys do not expire. Signatures do expire. How long a signature > is good for depends upon the software generating the signature, some > lets you specify. ldns I believe defaults to 60 days but I am not sure. > > The keys are in DNSSKEY records that are signed by your Key Signing > Key and must be resigning before the signature

I'm having problems getting my aastra 480i to register with the asterisk server. I can inititate calls from the phone, but >sip show peers does not show any IP address registered for this phone. I am probably missing something stupidly simple. Anyone have an example config to share or corrections for my configuration? Asterisk 1.2 aastra 480i CT has the 1.4 firmware <sip.conf>

hi, I have a really strange problem which is driving me crazy for days now. If I register my asterisk (tried all versions from 1.6 up to 13.x) with one sip registrar, everything works... calls go out and call come in... no 32 seconds limit. but as soon as I configure another sip registration on another server, outgoing calls drop after 32 seconds. as far as I know, there is no firewall in

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

Hi all, I've been pulling my hair out for two days over this problem... I did *a lot* of Googling around, I searched the list archives to no avail - no one has the same problem! I have two Avaya 4610sw phones. I installed the latest SIP firmware using the TFTP server. So far everything looks good. Each time the phone boots, it retrieves the 46xxsettings.txt from the TFTP server. My problem

Hi, I've just read the postings regarding the interworking between * and SER. As these persons seem quite knowledgeable on this, I would like to have their advise on my planned installation: - I have broadband cable access - I plan to install a SIP-aware router - I plan to install a Linux server with Digium analog IF card(s) for connection to my analog line (incoming and outgoing) - I plan

Hello, Our university has to upgrade soon its old Nortel PBX's which holds around 10,000 extensions tied to 5 PBXes. Up to now we thought about commercial solutions but now there is a window openning for open source solution. However, I need examples to convince that this solution is feasible, and preferably at other universities. Are there any pointers for such installations?

I am curious if it is advisable to use implement Asterisk as a Session Border Controller for a VoIP reseller environment. Users will terminate calls SIP to my server, which will authenticate them via RADIUS, perform a LCR lookup, select an appropriate trunk (based on LCR), and terminate the call (update RADIUS accounting at end of call). All while acting as a B2BUA to prevent the users from seeing