search for: sip_custom

Displaying 17 results from an estimated 17 matches for "sip_custom".

2008 Dec 01
2
Inbound calls from Asterisk to Asterisk with SIP "Forbidden" from '"asterisk"
...tcode=5260477782 amaflags=billing context=Incoming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpsetup=no t38pt_udptl = yes Asterisk 2 sip.conf GNU nano 1.3.12 File: sip_custom.conf [VoipDirect777821] type=friend host=141.122.139 username=VoipDirect777821 secret=wsPiOov8830 accountcode=5260477782 amaflags=billing context=Incomming disallow=all allow=g729 ;allow=alaw ;allow=ulaw trunk=no qualify=yes qualifysmoothing=yes nat=no canreinvite=yes dtmfmode=rfc2833 ;directrtpse...
2008 Apr 04
1
Friday April 4th @ 12 Noon EDT: VoIP Users Conference (Asterisk!)
Yes, we mostly talk about asterisk, hardware, software, phones, people, events and things asterisk-related. Asterisk is a registered trademark of Digium. Asterix is a registered trademark of Ren? Goscinny. We never discuss that, though. Here's the short URL for sending out to others that might be interested: http://x2z.eu It lands here: http://www.VoipUsersConference.org Once again, I
2009 Aug 14
2
no ring tone
...n't ring. Edit sip_nat.conf for proper NAT: localnet=192.168.1.0/255.255.255.0 externhost=pbx.DOMAIN.com (Set your external hostname name here) externrefresh=10 fromdomain=DOMAIN.com (Set your external domain name here) nat=yes qualify=yes canreinvite=no Add extra codecs to /etc/asterisk/sip_custom.conf allow=gsm allow=h261 allow=h263 allow=h263p videosupport=yes _________________________________________________________________ Windows Live?: Keep your life in sync. http://windowslive.com/explore?ocid=PID23384::T:WLMTAGL:ON:WL:en-US:NF_BR_sync:082009 -------------- next part -------------- An...
2005 Aug 16
1
problems with eyebeam - video phone
...dress to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ; allow=h263p ; H.263p is the enhanced video codec context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf And I left only H.263 basic in codec's configuration in Video Phone. No chance to get the communication in H.263 protocol. I saw that to use H.263+ protocol I need Asterisk CVS. I am not using asterisk CVS I am using asterisk 1.0.9 (last stable version a coup...
2010 Feb 02
0
Issue when reloading
.../etc/asterisk/sip_general_custom.conf': == Found == Parsing '/etc/asterisk/sip_nat.conf': == Found == Parsing '/etc/asterisk/sip_registrations_custom.conf': == Found == Parsing '/etc/asterisk/sip_registrations.conf': == Found == Parsing '/etc/asterisk/sip_custom.conf': == Found == Parsing '/etc/asterisk/sip_additional.conf': == Found == Parsing '/etc/asterisk/sip_custom_post.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found == Parsing '/etc/asterisk/phoneprov.conf': == Found -- Reloadi...
2011 Jul 13
1
Connect Avaya to Asterisk PBX
...t=1720 callerID="ALT Asterisk PBX" progress_setup=8 progress_alert=8 disallow=all allow=all dtmfmode=inband faststart=yes context=internal [Avaya] type=friend context=internal host=10.1.129.247 port=1720 canreinvite=no disallow=all allow=alaw dtmfmode=inband *Here is also the content of sip_custom.conf:* [general] context=internal videosupport=yes allow=h261 allow=h263 allow=h263p bindaddr=10.1.129.231 srvlookup=yes conreinvitte=no [1000] type=friend secret=malvin123 host=dynamic dtmfmode=inband disallow=all allow=all nat=yes Thanks & regards, Malvin -------------- next part ---------...
2005 Jun 01
0
Segmentation Fautl / Core Dump with G.729
...t = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=alaw allow=g729 allow=g723 context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown language=es #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf ---------------------------------------------------------------------------- --- SIP_ADDITIONAL.CONF ---------------------------------------------------------------------------- --- [as5300] type=peer qualify=yes host=xxx.xxx.xxx.xxx (AS5300 box) ----------------...
2010 Apr 30
0
Problems with t38modem and bitrate sent to t38-termination service
...9;t know if this is an Asterisk issue or a t38modem issue. My configurations are: /etc/asterisk/udptl.conf: [general] udptlstart=4000 udptlend=4999 udptlchecksums=no T38FaxUdpEC = t38UDPRedundancy T38FaxMaxDatagram = 400 udptlfecentries = 3 udptlfecspan = 3 use_even_ports=no /etc/asterisk/sip_custom.conf: [t38modem-options](!) type = friend host = 127.0.0.1 context = fax-out canreinvite = no disallow = all allow = ulaw t38pt_udptl = yes t38pt_rtp=no t38pt_tcp=no dtmfmode = rfc2833 nat = no qualify = yes [T38modem0](t38modem-options) port = 6060 [T38modem1](t38modem-options) port = 6061 [T3...
2005 Jul 06
1
SIP/2.0 403 Forbidden
...roblems. [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf [1000] username=1000 secret=abc123 context=mytest host=dynamic ----------------- /etc/asterisk/extensions.conf : [general] static=yes writeprotect=yes ;Suport phones ;SUPPORTPHONES=SIP/2205&SIP/2206&SIP/2207&SIP/2208&SIP/2209 [globals] XLITE=SIP...
2007 May 21
3
Aastra MWI
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the web interface by setting "Explicit MWI Subscription" to true, but no lights, no stutter tone. Firmware: 1.4.0.1048 Thanks! -- Warm Regards, Lee
2005 Aug 20
0
Help needed receiving incoming calls.
2010 Mar 09
0
DUNDI Sip authentication failure
...nounsolicited,nocomunsolicit,nopartial ;priv => dundi-priv-via-pstn,400,SIP,192.168.199.21/${NUMBER},nopartial [00:40:48:B2:78:6B] model = symmetric host = 192.168.199.23 inkey = 192.168.199.23 outkey = 192.168.199.21 include = priv permit = priv qualify = yes order = primary */etc/asterisk/sip_custom.conf language=fr nat=never ;Subscribecontext=ext-local [priv] type=friend dbsecret=dundi/secret context=dundi-priv-local host=192.168.199.23 qualify=yes* /etc/asterisk/extensions_custom.conf [ext-local-custom] ;for Direct IVR dialing if IVR is installed on the PBX B exten => _36X,1,Macro(dund...
2005 Jun 24
4
UTStarcom F1000 WiFi IP Phone Review
I bought a UTStarcom F1000 WiFi IP Phone from http://www.luxoncomm.com and tested it with Asterisk. This is a my first impression of the device. The F1000 supports SIP. It looks and operates like a cell phone, and connects to the Internet through WiFi, so you can use it at any WiFi hotspot. I set up a 802.11b wi-fi network with a Linksys BEFW11S4 Wireless-B broadband router with no security
2006 Apr 26
2
Unable to accept incoming PSTN calls
...e: [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf #include additional_a2billing_sip.conf extensions.conf: zapata.conf file: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Wh...
2006 Jun 09
3
FXO registration and VegaStream
I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
...closest proxy server host=proxy.mia.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3055741428 secret=MYSECRETPASS context=from-broadvoice ;Disable canreinvite if you are behind a NAT canreinvite=no ;Don't try to authenticate on incoming calls insecure=very #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf +++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EX...
2012 Jul 05
7
FreePBX: using context other than the default context and the generation for the configuration
Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I