search for: record_out

.... Seems it should work but doesn't. The Accounts are unreachable. Any Ideas? Main PBX: [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 [2001] username=2001 type=peer secret=2001 record_out=Adhoc record_in=Adhoc qualify=yes nat=no host=10.1.1.3 dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callerid=device <2001> allow=ulaw port=5061 [2002] username=2002 type=friend secret=2002 record_out=Adhoc record_in=Adhoc qualify=yes nat=never host=10.1.1.3 dtmfmode=rfc...

Hi all, having performance problems with various SIP-Phones, the manufacturer adviced us to add these parameters in sip.conf - unfortunately, neither one of us has an idea what these are supposed to do. I've seen various configuration files (sip.conf, iax.conf) posted on the net or this list using said paramters, but they seem to completely lack documentation (or is it just me?). Grepping

...00 (TCP and UDP) 5060 (TCP and UDP) I am using asterisk@home 1.3 , and xlite as softphone. I have tried to set the softphone I have set the extention parameters(in sip.conf) to: ;; Location A [200] username=200 type=friend secret=1234 record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never mailbox=200@default host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Location A" <200> ;; Location B [201] username=201 type=friend secret=1234 record_out=On-Demand record_i...

...horization/Authentication? I am not sure if this can be done with an entry into the sip.conf file, or by other means. My sip.conf file is shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device <201> [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 con...

...h accounts!!! -- Registered SIP '500' at 192.168.100.188 port 5060 expires 120 -- Registered SIP '400' at 192.168.100.188 port 5060 expires 120 how can i avoid this? i want to register only SIP 500!! this is a piece of my sip.conf [400] username=400 type=friend secret=400 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=400@device host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid= <400> My user [500] username=500 type=friend secret=500 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=500@device host...

...ral] bindport = 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 tos=0x68 notifyringing=yes notifyhold=yes limitonpeers=yes [120] type=friend secret=secret record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes mailbox=120 at default host=dynamic dtmfmode=inband disallow= dial=SIP/120 context=from-internal canreinvite=no callgroup= callerid=device <120> allow= accountcode=...

...the two. I have named each box asterisk1 and asterisk2. Does anyone have some working SIP and/or IAX trunk configurations they can send to me? Here is my current SIP config which doesnt seem to work: sip.conf on asterisk1: register=ast1:****@x.x.x.x [100] username=100 type=friend secret=**** record_out=Never record_in=Never qualify=no port=5060 host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="Ext 100" <100> [ast2] type=user secret=**** context=local [astrx2] username=ast1 type=peer secret=**** host=x.x.x.x sip.conf on asterisk2: register=ast2:****@...

...end of this mail. I am probably missing something very stupid, and I would like to hear what do you think it is. [channels] language=en threewaycalling=yes transfer=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes ;echotraining=800 faxdetect=incoming ;;;;;;[401] signalling=fxo_ls record_out=Adhoc record_in=Adhoc mailbox=401 immediate=no group=5 echotraining=yes echocancelwhenbridged=no echocancel=yes context=from-internal callprogress=no callerid=ZAP channel 1 <401> busydetect=no busycount=7 channel=>1 ;;;;;;[402] signalling=fxo_ls record_out=Adhoc record_in=Adhoc mailbox=40...

...est algorithm=MD5,realm="asterisk",nonce="62a12192" Content-Length: 0 I have defined two users in my sip.conf file as shown below: ; do not edit this file, this is an auto-generated file by freepbx ; all modifications must be done from the web gui [201] type=friend secret=201 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device <201> [202] type=friend secret=202 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes host=dynamic dtmfmode=rfc2833 dial=SIP/202 con...

Hello, I am trying to get to Jain Sip softphones to call one another via an Asterisk server. When I call from phone 1 to phone 2 there is audio transmission both ways, but when I call from phone 2 to phone 1 I don't get audio transmission and reception both ways. When I look at the asterisk log file it has an entry which says: "Oooh, format changed to 2". Would anyone know why

Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem. For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host" I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942

...all allow=ulaw allow=alaw context=default insecure=very qualify=yes nat=yes host=plasma.digitalvoice.ca register=XXXXXXXXXX:xxxxxx@plasma.digitalvoice.ca/franv [ext1] username=ext1 host=dynamic fromuser = XXXXXXXXXX authname= XXXXXXXXXX fromdomain = plasma.digitalvoice.ca type=friend secret=secret record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never dtmfmode=rfc2833 context=internal canreinvite=no insecure=very callerid=XXXXXXXXXX <xxxxxx> [Digital_out] type=peer secret=xxxxxx username=XXXXXXXXXX host=plasma.digitalvoice.ca fromuser=XXXXXXXXXX fromdomain=plasma.digitalvoice.ca i...

...n,Hangup exten => 503,hint,SIP/503 exten => ${VM_PREFIX}503,1,Macro(vm,503,DIRECTDIAL) exten => ${VM_PREFIX}503,n,Hangup ; end of [ext-local] *************************** 2 ************************** SIP_additional.conf one of my extension is configured as -- [507] type=friend secret=1234 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 nat=yes mailbox=507 at device host=dynamic dtmfmode=rfc2833 dial=SIP/507 context=from-internal canreinvite=no subscribecontext = ext-local notifyringing = yes callerid=device <507> ******************************************** 3 ********************...

Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor) suggested I add the following line to my sip.conf file: exten => 2060,3,Monitor(wav,myfilename) I can see how this could work, but I'm not sure what to replace "206...

I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}

...0 conference users Aug 2 04:03:46 VERBOSE[1552]: -- Registered channel 1, PRI Signalling signalling Aug 2 04:03:46 DEBUG[1552]: Updated conferencing on 2, with 0 conference users Aug 2 04:03:46 VERBOSE[1552]: -- Registered channel 2, PRI Signalling signalling Aug 2 04:03:46 WARNING[1552]: Ignoring record_out Aug 2 04:03:46 WARNING[1552]: Ignoring record_in Aug 2 04:03:46 WARNING[1552]: Ignoring echocancelwhenbridge Aug 2 04:03:46 ERROR[1552]: Signalling requested is FXO Kewlstart but line is in PRI Signalling signalling Aug 2 04:03:46 ERROR[1552]: Unable to register channel '1' Aug 2 04:03:46 W...

...co.com="30006" Supported: (null),X-cisco-xsi-6.0.2 Content-Length: 0 Expires: 3600 These messages repeat again and again. It does not look like the "SIP/2.0 200 OK" message is any better than 401 before. My config in sip_additional.conf is : [2001] type=friend password=2001 record_out=Adhoc record_in=Adhoc qualify=yes port=5060 pickupgroup= nat=yes mailbox=2001 at device host=dynamic dtmfmode=rfc2833 disallow= dial=SIP/2001 context=from-sip canreinvite=no callgroup= callerid=device <2001> allow= accountcode= call-limit=50 My updated SEP<MAC> file for this hard pho...

Have asterisk@home 1.2.1 The server is on an internal network eg 10.10.10.10 It is NAT'd 1:1 via Checkpoint firewall to external public IP eg 50.50.50.50 The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on extension 1055. Outbound calls to 1055 work perfectly. Inbound calls from 1055 get picked up as if it were an external call (see below) and goes straight to the ring

...s the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed - Unauthorized) Any ideas? Please find below my conf files. SIP.CONF [300] username=300 type=friend secret=cisco record_out=On-Demand record_in=On-Demand qualify=no port=5060 nat=never mailbox=300@default host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no callerid="" <300> SIP000CCE351C07.cnf # SIP Configuration Generic File (start) # Line 1 Settings line1_name: "300"...

...y input/suggession on this. Obaid. Here are my conf files, followed by SIP debug output on asterisk. trunk 4= SIP trunk 24.XX.XXX.101 ---> Asterisk server on Public IP 209.XXX.XXX.113 ---> SIP gatway ---------------iax_additional.conf-------------- [20] username=20 type=friend secret=XXX record_out=On-Demand record_in=On-Demand qualify=no notransfer=yes mailbox=20@default host=dynamic context=from-internal callerid="512538XXXX" <20> -------------------Sip_additional.conf--------------- [23] username=23 type=friend secret=XXX record_out=On-Demand record_in=On-Demand qualify=...