search for: sip_additional

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

...you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension defined the line callerid=device <567>. If you look at the mysql tables, the only place where the field outbound CID you entered is recorded is in the users table. If you manually edit etc/asterisk/sip_additional.conf and adjust the line callerid=wh...

...o advance further. The firmware was able to install smoothly but I am stuck at the registration part. I went through another post here on this subject at : http://forums.digium.com/viewtopic.php?t=15212&highlight=7970 This helped me get past the SIP 401 Unauthorized error when I went into the sip_additional.conf file and changed the "secret=" line to "password=" However, the phone is still stuck in Registering, and I see these new messages on the asterisk CLI : <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 10.16.121.1...

...; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) externip = 62.219.XXX.XXX disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown nat = yes #include sip_nat.conf #include sip_additional.conf [root@crystalclear root]# cat /etc/asterisk/sip_additional.conf register=TollIPdemo1:somesecret@sipdevice.FQDN.net [sip-devices] username=TollIPdemo1 type=friend secret=somesecret host=sipdevice.FQDN.net fromuser=TollIPdemo1 context=from-pstn canreinvite=no callerid=TollIPdemo1 Any informa...

Hi, I'm running *@home 0.5 exactly as it comes off the .iso image and have configured an extension (206) using AMP for a grandstream 102. Have checked sip_additional.conf looks ok, but I can't get the incoming cli on the 102 to read anything other than 't ri', apparently from other posts the phone is trying to display 'asterisk'! I've got a couple of pc's running x-lite and an ata 486, all can call each other and the x-lites disp...

...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) ---------------------------------------------...

Greetings. I'm running asterisk here (elastix) and I have a few extensions configured in it. I have here two different callgroup/pickgroup where the extensions are configured in, but it doesn't work when I try do pickup a call. Looking the config file (sip_additional.conf) I see they are not configured with callgroup/pickgroup, the fields are empty. Manually inserting callgroup/pickgroup on the extensions worked just fine but the next day the configuration just vanished and the extensions was not working. Has someone a clue of whats going on here? -- Att.*...

...resses 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 couple of week ago..) Is there a...

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

...isk, my DID phone will work with stanaphone (in which i'm crapping in my pants if they'll exist cause they never return emails). Though CLI won't work. if i do '/var/run', my DID won't work, but CLI will... I've tried just coping over the extensions_additional.conf and sip_additional.conf files from my old setup to my new one, and that didn't work. Maybe I should just install my previous version. Are there QoS differences though? I'd rather not regress if that were the case. -- Anything else, let me know. - Dominic "It is not the force of a stroke that makes...

...a Polycom phone at your disposal. I ran this test as under: 1) Configure a Polycom Phone as SIP extension under Asterisk@home 2) Try to make a call - It will not register with Asterisk as asterisk@home does not use this option in it's sip configuration and even if you add this manually in the sip_additional.conf, AMP will overwrite(remove) this line from the sip_additional.conf To make Polycom phone work with Asterisk, add the sip.conf entry into the main sip.conf, using progressinband=no option. Polycom phone will immediately connect. Seshu -----Original Message----- From: asterisk-users-bounces@l...

...blems. [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_additional.conf register => xxxxxxxxxx@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@sip.broadvoice.com/2197 [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=xxxxxxxxxx secret=pppppppppp username=xxxxxxxxxx insecure=very context=from-broadvoice authname=...

...n => ${VM_PREFIX}502,n,Hangup exten => 503,1,Macro(exten-vm,503,503) exten => 503,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=...

...ng '/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 -- Reloading module 'res_odbc' (ODBC resource) == Parsing '/etc/a...

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

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,Mo...

...ght now, "sip show registry" shows this: Host Username Refresh State sip.broadvoice.com:5060 6033751414@s 120 Auth. Sent I added this to my sip.conf: externip=xxx.xxx.xxx.xxx localnet=10.1.0.0 localmask=255.255.0.0 nat=yes My /etc/asterisk/sip_additional.conf contains this relevant portion: register=> nnnnnnnnnn@sip.broadvoice.com:pppppppppp:nnnnnnnnnn@sip.broadvoice.com/2 00 [from-broadvoice] username=nnnnnnnnnn user=nnnnnnnnnn type=user secret=pppppppppp nat=yes insecure=very host=sip.broadvoice.com fromdomain=sip.broadvoice.com dtmfmode=inb...

...s certainly an error in my sip.conf but I can't find where it is... I'm sure about IP addresses. SIP.conf [general] port = 5060 bindaddr = 213.177.xxx.yyyy ; My Pubic Address. context = incoming_SIP disallow=all allow=alaw allow=gsm allow=ulaw #include sip_additional.conf Extensions.conf exten => 4700,1,Dial(SIP/4700@213.177.xxx.yyy|30|m) Adress to Contact... Please help Regards, Ignace

...Port to bind to (SIP is 5060) bindaddr = 202.69.190.244 ; 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_additional.conf If there are some parameters that i hould define please let me know... if this work all the configuration will be posted as reference for others... THank you all. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipe...

...et mask) nat=yes 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_additional.conf register={userid}:{password}@{domain}.net [200] username=200 type=friend secret=??? qualify=no port=5060 pickupgroup= nat=never mailbox= host=dynamic dtmfmode=rfc2833 disallow= context=from-internal canreinvite=no callgroup= callerid="Demo" <200> allow=...