similar to: sip.conf outboundproxy

Is it possible to load phone registration information stored in sipfriends MySQL DB, so that Asterisk "thinks" those phones are already registered? This would be very usefull for a redundant server... Regards, Ricardo Carvalho. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Thanks a lot for the help. But if Asterisk has private IP address and the only way to access it from remote sites is to have vpn connection to the site that asterisk existed (the site has vpn), then how that will happen from the Mobile to be able to run the softphone from the mobile? Any help? Regards Bilal ----------------- I installed out of curiosity today, and guess what? You can do SIP over

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment

I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the "host=" IP, instead of the "outboundproxy=" IP as it should, verified with tcpdump. I've done a

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug.

Dear all, I'm trying to configure Asterisk to be able to ask the caller to enter a given password in order to continue dialplan execution. I've tested this feature using the Authenticate application like this: exten => _X./5219,1,Answer exten => _X./5219,2,Authenticate(1234,a) exten => _X./5219,3,Playback(pin-number-accepted) exten => _X./5219,4,Dial(SIP/${EXTEN},120)

I've tried to use a Cisco power injector to supply power over Ethernet to a GXP2000 phone without success. Although when I plugged these phone to a PoE capable Cisco Switch it worked without a problem! Knowing that all these three equipments implement IEEE 802.3af protocol, why doesn't it work with the Cisco power injector? Anyone also had this problem before? Thanks, Ricardo Carvalho.

Is it possible to implement in the Asterisk dialplan some way to authenticate a user with a dialed passcode which opens session that stays active enabling the user to make and receive calls, until the user logs off with another dialed passcode? I am aware of the Asterisk application 'Authenticate', but as far as I know, with this application the user meeds to dial his pin at each call he

How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication

Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem

Hi, How can I configure my asterisk server so that I can receive incomming calls comming from the same IP from where my server also receives phone registrations? The problem is that since the moment any extension registers at that IP (actually I have a registration proxy running at that IP), asterisk no more accepts calls coming from a SIP trunk I also have at that IP, replying back with

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Hi, I manage an SBC which stands between my company server farm and some SIP telco trunks. The system works fine, for inbound and outbound calls. Now I've configured the SBC to also act as a registration proxy, forwarding SIP registrations coming from the Internet to my asterisk servers. It all seems fine, but it doesn't work well, because by the time at least one phone registers through

How can I match wildcards inside a GoToIf? I have something like this, but it doesn't work: [default] exten => _2XXXXXXXX,1,Macro(outcall,${EXTEN}) [macro-outcall] exten => s,1,GotoIf($["${ARG1}" = "220408XXX"]?2:3) exten => s,2,Hangup Any ideas? Regards, Ricardo.

Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some adjustments to the configuration files to port it to the new version, calls between registered phones in asterisk, work fine, but inbound calls coming from the SIP trunk I have with a telco to asterisk, don't work anymore. I don't know why!... This is the SDP portion that comes in the INVITE messages of calls

Hi all, In my server dialplan, it first tries to dial possible SIP URI contacts returned by DNS lookups using the ENUMLOOKUP function; it only sends calls to PSTN when there aren't any NAPTR records of the dialed number. Problem arises when my Internet connection is down to some locations, which inhibits my Asterisk server to reach the DNS servers to do those lookups. In those cases, calls

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of

Dear all, I've noticed that when I have a phone registered in Asterisk, and then I register another phone with the same user, the "sip show peers" in the CLI shows that Asterisk replaced the IP of the first phone by the IP of the last one registered for that user. Consequently, if someone calls that user, only the last phone rings!! How may I configure Asterisk to be able to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom