search for: sipprovid

...king, but I have not seen any documentation of these setups. So far, SIP Clients can talk to each other. I can also start a call through Asterisk to a VoIP provider, but there is a problem after the first ring: Here is the output: -- Executing Dial("SIP/-08114560", "SIP/13239381067@SIPprovider") in new stack -- Called 13239381067@SIPprovider -- SIP/SIPprovider-5e0c is making progress passing it to SIP/-08114560 -- SIP/SIPprovider-5e0c answered SIP/-08114560 -- Attempting native bridge of SIP/-08114560 and SIP/SIPprovider-5e0c I have tried this with my SIP client behind a NAT and...

...d for connect to my sip provider. My sip.conf is: [general] ;disallow=gsm ;allow=ulaw port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls callerid=No CallID register => user:password@sipprovider.com [sip_proxy] type=friend username=user fromuser=user secret=password host=siprovider dtmfmode=inband The problem is: If i put in the [sip_proxy] section type=friend, incoming calls doesn't works. If the type is set to another value (for example peer) incoming calls works fine, but outg...

Sorry for the simple question. I'm trying to match "sipprovider.nocredit" but the following doesn't execute NoOp (it runs "context" but not "context-custom"). What am I doing wrong? [context] include => context-custom exten => _.,1,Set(GROUP()=1) exten => _.,n,Goto(destcontext,${EXTEN},1) [context-custom] exten => si...

Hi, I am diverting an incoming call to a mobile phone and a landline using the following:- exten => 0203000000,3,Dial(SIP/442080000000 at sipprovider&SIP/44700000000 at sipprovider,120,r) For billing purposes, i need to be able to work out whether the diverted call was answered by the mobile or whether it was answered by the landline. The CDR only shows the full Dial() information, and not what the final destination was. lastdata contai...

Hi, i am from Germany, so excuse my School English. I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update of Asterisk 2 wooks ago, Echos accure in my SIP Calls. I use SNOM 360, sometimes there is no echo (for example if i call myself via SIP->Asterisk->SIPProvider->TELEKOM->ISDN) but if i call other people there occures Echo many times. The Routing is always the same : SIP (SNOM) -> Asterisk -> VoIPProvider -> ISDN/POTS Can i control the cancellation with the zapata.conf ? I have tried this with "echocancel=..." and so on, no lu...

...isk Open SOurce PBX" Hangup is executed. This seems to be the 'default' behaviour of Asterisk and also does not seem to be right :-) Here is what asterisk shows on the console when a normal BUSY number is dialed: -- Executing Dial("SIP/4077-030d", "SIP/17346621122@sipprovider-out") in new stack -- Called 17346621122@sipprovider-out -- Got SIP response 486 "Busy Here" back from 216.164.114.122 -- SIP/sipprovider-out-c3fd is busy == Everyone is busy/congested at this time (1:1/0/0) -- Timeout on SIP/4077-030d == CDR updated on SIP/4077...

...reated by 'pbx_config' ] '123' => 1. Answer() [pbx_config] 2. AGI(/usr/local/share/examples/asterisk/agi/agi- test.agi) [pbx_config] 3. Hangup() [pbx_config] '_X.' => 1. Dial(SIP/sipprovider/${EXTEN}||L[300000:240000:30000]) [pbx_config] 2. Hangup() [pbx_config] I am using 1.2.17. /usr/local/sbin/asterisk -vvvvvv -g -dddddd -c Does not show anything to even indicate * is trying anything unusual with regards to limits or warnings. It just seems to ignore the di...

...username . Now... When I put this in the sip.conf file I have found that Asterisk is not able to parse it correctly and instantly goes to the email server to authenticate the sip user upon registration Here is the line below in my sip.conf file register => username@theiremailhost.com:password@sipprovider.com THe error is below Aug 16 11:30:05 NOTICE[114695]: chan_sip.c:3922 sip_reg_timeout: Registration for 'john@useremail.com@sip.voipamericas.com' timed out, trying again Aug 16 11:30:06 NOTICE[114695]: chan_sip.c:6575 handle_response: Failed to authenticate on REGISTER to '&lt...

...ing from scratch and all works great except I cannot make an outbound sip-to-PSTN call and do not fully understand how to configure it. I've been folowing some examples and keep running into this stumbling block: As soon as I add (to sip.conf) this section: [siprovider.com] type=peer host=sipprovider.com fromuser=2135551212 secret=2135551212 authname=2135551212 fromdomain=siprovider.com I no longer can recieve ANY inbound calls from the PSTN via my sip provider. I've tried many variations of attempting to get this section (I think it's referred to a 'sip channel) into my sip....

...voicemail. The dial command is: exten => s,1,Dial(ZAP/1/6135551111,15) exten => s,2,VoiceMail(u1) exten => s,102,VoiceMail(b1) The call will continue to ring beyond 15 seconds. What's interesting is that a SIP channels does not have this problem. exten => s,1,Dial(SIP/6135551111@sipprovider,15) exten => s,2,VoiceMail(u1) exten => s,102,VoiceMail(b1) I have tested in Asterisk 1.2.7.1 and 1.2.8, both have a problem with the Zap channel. Any ideas? TIA, -Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asteri...

Hi, it seems that there is a bug in asterisk's dtmf mode autodetection. Assume following sip.conf: [sipprovider] disallow=all allow=g726 dtmfmode=auto DTMF does not work. It seems rfc2833 mode is chosen despite it being obvious that this cannot work! The following configuration is necessary to get DTMF to work: dtmfmode=info In my opinion, this behaviour is counter-intuitive. I am using asterisk 1.2....

.... $AGI->set_variable('DAILNO', $BranchPhone); $AGI->exec("Macro","agidial"); And my macro: [macro-agidial] exten => s,1,AGI(getcid.pl,${CALLERID(NUM)},1) exten => s,2,NoOp(DIALNO=${DIALNO}) exten => s,3,Dial(SIP/${DIALNO}@SIPPROVIDER,60) exten => s,4,GotoIf($["${DIALSTATUS}" = "CONGESTION"]?10) exten => s,6,Hangup() exten => s,10,Dial(IAX2/SERVER2/${DIALNO}) exten => s,12,Hangup() but when the macro is called, Dialno = nothing. -------------- next part -------------- An HTML a...

...versations : [default] exten => s,1,NoOp(call from 3StarsNet) exten => s,n,Dial(SIP/grandstream,30) I would like : [from3starsnet] exten => s,1,NoOp(call from 3StarsNet) exten => s,n,Dial(SIP/grandstream,30) Problem 2 Setup : Grandstream --> Asterisk --> Endian_Firewall --> SIPprovider Problem : Called party can not here me (I'm on the Grandstream) while I can here the other side clearly (GSM/cell phone number). Making a call or receiving a call makes no difference. Configuration Asterisk : rtp.conf : rtpstart=11000 rtpend=11500 firewall : -A RH-Firewall-1-INPUT -p udp...

...idged at ringing. So, this can confuse the callee. So, I thought I should send calls to a context first and then ask customer enter $spoofNumber and then place call but then I am facing another problem. Using that, the internal context is called first and all announcements are made and then the SIP/sipProvider/$phoneNumb is dialed. Or at least it's dialed at the same time but since it takes time to pick ones phone context already goes over it's announcement for putting in spoof number and dialnumber. Please guide me how to do this properly. Following is the code and the context: $sys_ip = &quo...

...eve,1) ;exten => 2,1,Goto(default,mark,2) [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include => demo ; ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. ; Note that you must have a [sipprovider] section in sip.conf whereas ; the otherprovider.net example does not require such a peer definition ; ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) ; Real extensions would go here. Generally you want real e...

...eve,1) ;exten => 2,1,Goto(default,mark,2) [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include => demo ; ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. ; Note that you must have a [sipprovider] section in sip.conf whereas ; the otherprovider.net example does not require such a peer definition ; ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) ; Real extensions would go here. Generally you want real e...

...teve,1) ;exten => 2,1,Goto(default,mark,2) [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include => demo ; ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. ; Note that you must have a [sipprovider] section in sip.conf whereas ; the otherprovider.net example does not require such a peer definition ; ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) ; Real extensions would go here. Generally you want real e...

...ode of the call and if out-of-band is not supported, can you change it to inband as a last resort? Is there a way to set priority for DTMF signalling like you can do with codecs? I have tried that (see below) but it seems to default to inband (is this even a proper way to handle 2 DTMF modes?). [sipprovider] type=friend host=xxx.xxx.xxx.xxx disallow=all allow=ulaw maxexpirey=15 dtmfmode=rfc2833 dtmfmode=inband nat=no insecure=very canreinvite=no I have searched and searched and the closest thing that I have found is "SIPDtmfMode" but from what it looks like it needs to be initiated befo...

Hello list ! SETUP : Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone PROBLEM : I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing. INFORMATION : This is the output on the CLI of the local Asterisk-server : [Oct 3 17:40:33] -- Executing [0473775006 at intern:1] NoO...

...ve,1) ;exten => 2,1,Goto(default,mark,2) [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; include => demo ; ; Extensions like the two below can be used for FWD, Nikotel, sipgate etc. ; Note that you must have a [sipprovider] section in sip.conf whereas ; the otherprovider.net example does not require such a peer definition ; ;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,,r) ;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) ; Real extensions would go here. Generally you want real...