search for: provider1

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call: Num...

Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send Dial(SIP/number|10|m(number)) I have silence on the line. No ringtone, nothing. Now contacting a friend whose Asterisk is connected to another provider (let's give hi...

...Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at lists.digium.com > > On 14.12.2017 16:30, basti wrote: > Hello, > I am new on asterisk and do some tests on freepbx. > > I have 2 SIP provider: > > Provider1: In-/Out- Flatrate, only 1 Number > Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers > > On Asterisk site i have 3 phones > (branch ??, don't know how its called in asterisk) > > Is it possible to do something like: > > Phone 1: Incoming C...

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten => _1NXXNXXXXXX,100,NoOp(Calling my cell w/special CID) exten => _1NXXNXXXXXX,n,Set(CALLERID(all)="Dude" <5551112233>) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/...

...they're from a VoIP provider they are in e.164 format already, where they're from BT ISDN lines they're usually the last 6 digits of the number. For outbound calls, there's a context called [outboundpstn] containing the following entries: exten => _00.,1,Macro(outbound,{EXTEN},,provider1,provider2,pstn) exten => _0[12]XXXXXXXXX,1,Macro(outbound,${EXTEN},,provider1,provider2,pstn) exten => _07XXXXXXXXX,1,Macro(outbound,${EXTEN},,provider1,provider2,pstn) exten => _08[47]XXXXXX.,1,Macro(outbound,${EXTEN},,provider1,provider2,pstn) exten => _0[58]0XXXXXX.,1,Macro(outbound,...

...255.255.255.0 U 0 0 0 eth0 0.0.0.0 100.10.10.254 0.0.0.0 UG 0 0 0 eth1 Then I've setup two additional routing tables with iproute2 and linked to the main rule list like this # cat >> /etc/iproute2/rt_tables << EOF 10 provider1 20 provider2 EOF # ip route add 100.10.10.0/24 dev eth1 table provider1 # ip route add default via 100.10.10.254 dev eth1 table provider1 # ip rule add from 100.10.10.0/24 table provider1 # ip route add 99.10.11.0/24 dev eth2 table provider2 # ip route add default via 99.10.11.254 dev eth2 tab...

Hi I''ve just tracked down a wierd error that AFAICT is caused by an error not being raised in the setup: context "An Asset" do setup do @provider = Provider.create(:name => "Provider1") @product = Product.new(:name => "Product1", :provider => @provider) @applicant = Applicant.new(:first_name => "Fred", :last_name => "Flinstone", :email => "fred at flinstone.com") @asset = Asset.new(:value...

...pjsip show registrations No objects found. asterisk13*CLI> pjsip show contacts Contact: <Aor/ContactUri...................................> <Status....> <RTT(ms)..> ========================================================================================= Contact: provider1/sip:1XXXNNNYYYY at sip.provider1.com Unknown nan asterisk13*CLI> pjsip list aors Aor: <Aor..............................................> <MaxContact> ========================================================================================= Aor: p...

..... In extensions.conf, there are a number of contexts defined for each group of users, along the lines of: [groupa] [groupb] etc. In each of those, there's a command include => outbound [outbound] has entries similar to the following: exten => _0[123]XXXXXXXX,1,Macro(outbound,${EXTEN}, provider1, provider2) the macro "outbound" is defined in extensions.ael as follows: macro outbound (number, route1, route2) { dosomestuff; } This has worked fine in 1.2 and 1.4, but seems to be choking on 1.6. I've looked through the various changes.txt files, and have read mention of replac...

Hi, I put on sip.conf the following line #include "sip.d/*.conf" inside I have files like that provider1.conf provider2.conf But asterisk does not want to load it This is the error Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35 VERBOSE[8756]: == Parsing '/etc/asterisk/sip.d/*.conf': Not found (No such file or directory) this is the ls result...

...w occasions, packets are routed to the wrong interface. I''m not sure to understand rules and routes traversal correctly (I couldn''t find answers in the howto). So, here are my questions: 1. How does the rule traversal work exactly? If I have rules like this: 100: from all lookup provider1 200: from all lookup provider2 What does make stop the traversal? For instance, does a default route in provider1 would stop the traversal? 2. are tables entirely separated? Say, if you have this: 100: from all lookup table_part1 200: from all lookup table_part2 and table_part1 contains the route...

...ers. It is done as follows (once more building on the example in the section on split-access): ip route add default scope global nexthop via $P1 dev $IF1 weight 1 \ nexthop via $P2 dev $IF2 weight 1 "" What does it mean? It will split the traffic? Eg. some traffic goes through provider1 and some through provider2 ?? Or if provider1 goes down then the packet will go through provider2? I am asking it because I want to create an alternate default gateway when my ADSL link had down. Do you suggest any other solution? Thanks! Leonardo Borda Netwall Tecnologia e Projetos - http://...

Ok ive been trying to get this to work for about half a year now. Ive searched all over the internet for a solution for my problem. Ive found some solutions, but they only led me to yet more problems. What we want to do is the following: I live in a student complex with 7 other people. Every room has its own internet connection from the same ISP. Ip, gateway, subnet are asigned through dhcp on

...> > asterisk13*CLI> pjsip show contacts > > Contact: <Aor/ContactUri...................................> > <Status....> <RTT(ms)..> > > ========================================================================================= > > Contact: provider1/sip:1XXXNNNYYYY at sip.provider1.com > Unknown nan > > asterisk13*CLI> pjsip list aors > > Aor: <Aor..............................................> > <MaxContact> > > ===================================================================...

...> pjsip show contacts >> >> Contact: <Aor/ContactUri...................................> >> <Status....> <RTT(ms)..> >> >> ========================================================================================= >> >> Contact: provider1/sip:1XXXNNNYYYY at sip.provider1.com >> Unknown nan >> >> asterisk13*CLI> pjsip list aors >> >> Aor: <Aor..............................................> >> <MaxContact> >> >> ====================================...

Hi, I''m a bit stuck with mocking a unit test of a join model with regards to a validates_uniqueness_of clause. I have class Book < AR:B has_many :clippings various validates_presence_of etc end class Clip < AR:B has_many :clippings various validates_presence_of etc end class Clipping < AR:B belongs_to :book belongs_to :clip validates_uniqueness_of :clip_id,

Hello, I have the following setup: sip phones <->SER <-> asterisk <-> voip provider1 <-> voip provider2 i got a toll-free DID from voipprovider1 to allow people from outside to call into asterisk, get authenticated, and use voipprovider2 to call out (kind of a primitive calling card app). anyway, voiprovider is giving my bad DT...

Is there a way to override the fromdomain specified in the sip.conf and instead set the value from the dialplan? If we use: Set(CALLERID(num)=user at domain.com The SIP From header turns into: user at domain.com@10.10.10.10 We want user at domain.com, and we can't have an entry in sip.conf for every provider. -- Eric Chamberlain

Hi I want to be able to use one provider if I dial 0 before the number and another if I dial 1 before, how can I do that in asterisk 1.6? /ralf Ralf Tr?skman, IT AdLibris AB, Box 3667, 103 59 Stockholm. Bes?ksadress: Sveav?gen 56C, 111 34, Stockholm - Obs ny address! Dir: +46-(0)8-5460 60 91, mob: +46-(0)70-7548074, vxl: +46-(0)8-5460 60 00, fax: +46-(0)8-5460 60 99 ralf at

...lf a ring and then the hangup noise generated by the SIP device when a number they call is busy. Many thanks!! Chris PS please Cc: me a copy as well as to the list in case I miss it - Thanks. << extensions.conf >> exten => _00.,1,AbsoluteTimeout(3600) exten => _00.,2,Dial(${PROVIDER1}/${EXTEN:2},30,r) exten => _00.,3,Answer exten => _00.,4,Hangup exten => _00.,103,Dial(${PROVIDER2}/011${EXTEN:2},30,r) exten => _00.,104,Answer exten => _00.,105,Hangup <<iax.conf>> [iaxy] type=friend accountcode=iaxy disallow=all ;;allow=adpcm allow=ulaw username=iaxy...