search for: mysipprovid

...to a SIP Express router. Inbound calls to my asterisk server works just fine, but when i try to make outbound calls I get the following error message: Nov 27 22:40:48 NOTICE[4687]: chan_sip2.c:7967 handle_response: Failed to WWW-authenticate on INVITE to '"username" <sip:username@mysipprovider>;tag=as5399a078' I'm currently running Asterisk CVS-HEAD-11/27/04-15:07:27 and chan_sip2.c downloaded today. I had the problem using chan_sip as well, but the error message was slightly different. Below is a packet dump from the SIP session. My current sip.conf configuration is: -...

...k box with SER (which I don't begin to understand yet) at the office. In order to try to understand how all this works, I have stripped my extensions.conf down to almost nothing. I am building it up piece by piece. This is the entirety of my extensions.conf file: [globals] OUTBOUNDTRUNK=SIP/mysipprovider.com [from-internal] exten => 105,1,Answer() exten => 105,2,Playback(abandon-all-hope) exten => 105,3,Hangup() exten => 106,1,Dial(${OUTBOUNDTRUNK}/916xxx6000) exten => 107,1,Dial(${OUTBOUNDTRUNK}/916xxx2128) This is all just testing. When I dial 105 from either of my softphones,...

If you update your source from the CVS, you'll get a new SIP channel that supports a new syntax for SIP calls in extensions.conf If you define a SIP peer in sip conf, like [mysipprovider] ... You can now use dial(SIP/mysipprovider/extension) Where the part "mysipprovider" is related to the sip.conf section. Also, you can dial any SIP URL by dial(SIP/extension@domain) like dial(SIP/oej@edvina.net) In this case, "domain" referes to a Inter...

Hi, I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04). These two boxes talk to eachother via sip, not iax. Since the upgrade, I get the error "Failed to authenticate on INVITE" trying to make calls to/from either box. Removing the secret from each box's sip config seems to work but is utterly braindead. Has anyone seen this? - Eric

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

...m". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages and have noticed the following commented lines in sip.conf: ; ;register => 1234@mysipprovider.com ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. ; But I am a little confused how this should be implemented. -- Steve Woolley ADS Telecom, Inc. 59 Skyline Drive #1250 Lake Mary, FL 32746 Phone: (407) 682-6226...

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

...ti ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ;videosupport=yes ; Turn on support for SIP video ;disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ; ;register => 1234@mysipprovider.com ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 123 ; [sip7101] context=sip type=friend secret=blah auth=md5 ; defaultip=192.168.5.10 host=dynamic dtmfmode=inband mailbox=7101 -- Using M2, Opera's revolutionary e-mail...

...ncluding an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line register => <telephone number>:<password>@sip.myprovider.com in sip.conf. I also included a section [mysipprovider] type=peer context=fromINTERNET host=sip.myprovider.com and used Dial(SIP/<telephone number>@mysipprovider) in my dialplan. Context fromINTERNET only consisted of exten => s,1,Dial(FXSport/0,,tH) This setup was working great, but now I want to have the two RGs communicate directly to e...

...context = dummy deny=0.0.0.0/0 permit=<IP address of VOSP server> externip=<public IP address of NAT router> localnet=192.168.0.0/24 disallow=all allow=ulaw allow=alaw allow=gsm ;all RTP packets go through Asterisk canreinvite=no ;incoming calls from VOSP register => me:mypasswd at mysipprovider.com ;for outgoing calls to VOSP [vosp] ;friend = peer+user type=friend username=me fromuser=me fromdomain=mysipprovider.com authname=me secret=mypasswd host=mysipprovider.com insecure=very qualify=yes context=outgoing ;Since VOSP is on the Net, nat=no or nat=yes? nat=no ;extension for XLite [60...

Hi, the documentation of sip.conf is telling me this: ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension In reality it jumps to the extension 1234 in the context and not to s So it is much more complicate to write an proper dialplan. Is this an bug or is the documentation not up to date? best regards Thomas

...------------------------------------------------------------------------------------- [My sip.conf in Asterisk] [general] ; SIP Client Config - Start externip=*.*.*.* localnet=192.168.100.0/255.255.255.0 register => 1000:jrcyagi at 192.168.100.1/1234 ; To make a call to a external SIP server [mysipprovider-out] type=friend secret=jrcyagi username=1000 host=jrc.nwt.com fromuser=1000 fromdomain=jrc.nwt.com canreinvite=no insecure=very qualify=yes nat=yes context=from-mysipprovider ; is further defined in extensions.conf ; SIP Client Config - End ; SIP Server Config - Start [1000] type=friend secret...

...e type in NOTIFY ;videosupport=yes ; Turn on support for SIP video disallow=all ; Disallow all codecs ;allow=ulaw ; Allow codecs in order of preference ;allow=ilbc ;VoWLAN testing! allow=ulaw allow=alaw allow=gsm ;allow=all ; ;register => 1234@mysipprovider.com ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 ;mailbox=1234,2345...

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

Sorry not to have observed etiquet and lurked here for a bit before wading in with a question but I have an issue that may well be because I dont know enough about what asterisk is actually doing under the hood to understand why I cant do what I want with asterisk. Im hoping that someone can point me in the right direction :-) This is what I have: Mandrake 2006 running Asterisk 1.2.3 - no

...nstalled for testing. I don't have any extra hardware installed so far, was attempting to just try out connectivity. I am having some probs with the configuration, maybe someone out there can give me some tips : firstly on modifying the sip.conf file I got stuck at the line register => 1234@mysipprovider.com What exactly is a SIP provider? is this essential? Leaving the line as it was in the sample config file, asterisk crashes the machine after trying to read the SIP.conf (Crashes to the extent that the machine freezes .. ) What I would *like* the system to do is as follows : for now, just tak...

I have an Asterisk 1.4.2 server with 3 different SIP providers and Asterisk for Skype gateway installed. Periodically the SIP providers go offline for some reason, or the Skype connection fails. When this happens, I lose my SIP registration to the provider. Unfortunately I don't know this has happened until someone eventually contacts me to say, "I tried to call you but it

Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use "register=>......" on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions .... ? thanks Billy

...; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in NOTIFY ; ;register => me@mysipproxy.org ; Register with a SIP provider ;register => 2345@mysipprovider.com/1234 ; Register 2345 at sip provider as 1234 here. ; [snom1] type=friend host=129.26.10.121 dtmfmode=rfc2833 mailbox=2101 reinvite=no context=local callerid="Studenten 1" <2101> [snom2] type=friend host=129.26.10.122 dtmfmode=rfc2833 mailbox=2102 reinvite=no context=local ca...

...peer/user also ;notifyringing = yes ; Notify subscriptions on RINGING state GNU nano 1.3.8 File: sip.conf ; (provider). ; ; host is either a host name defined in DNS or the name of a section defined ; below. ; ; Examples: ; ;register => 1234:password@mysipprovider.com ; ; This will pass incoming calls to the 's' extension ; ; ;register => 2345:password@sip_proxy/1234 ;registertimeout=20 ; retry registration calls every 20 seconds (default) ;registerattempts=10 ; Number of registration attempts before we give up...