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...=> s,2,Dial(SIP/trunk-out/37052390920|60|rL(10000000000000)A(conf-enteringno)) But these prompts play not in the same time: just after conf-enteringno prompt asterisk plays hello world promt. -- <SIP/trunk-out-08155880> Playing 'conf-enteringno' (language 'en') -- <SIP/sip3.call.lt-08151550> Playing 'hello-world' (language 'en') So my question is , how to do this in the same time. Maybe somebody is using Dial G(context^exten^pri) for this purpose? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.d...

...ues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> -- Executing Wait("SIP/sip3-768a", "1") in new stack -- Executing AgentLogin("SIP/sip3-768a", "") in new stack Aug 12 16:31:37 DEBUG[1127562160]: rtp.c:1156 ast_rtp_write: Ooh, format changed from UNKN to ULAW Aug 12 16:31:37 DE...

...and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060 N 1112530146 105 Registered Mon, 08 Apr 2013 06:02:09 1 SIP registrations. Asterisk*CLI> Here is the dial plan: [incoming] exten => 17036361355,1,Playback(beep) exten => 17036361355,2,SayDigits(${EXTEN}) exten => 17036361355,3,G...

...me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ / \ User2 ------ Go to register ------- SIP2 ----- Whereis? --> DB / \ / User3 -- -- SIP3 -- Where users no matter who they are, register and are passed off to the next server in sequence... For example, ten people are all registering right now... User1 --> SIP1 User2 --> SIP2 User3 --> SIP3 User4 --> SIP1 And so on... where an ATA, VoIP phone, etc., would have its inform...

...r Username Refresh State Reg.Time 0 SIP registrations. Asterisk*CLI> my config is this: [outgoing] username=5552530146 (your VoIP VoIP account assigned while signing up) type=peer qualify=yes secret=iblockedthis (your VoIP VoIP password) nat=auto insecure=invite,port host=sip3.voipvoip.com fromuser=5552530146 (your VoIP VoIP account assigned while signing up) fromdomain=sip3.voipvoip.com dtmfmode=rfc2833 disallow=all allow=g729 allow=ilbc allow=ulaw allow=alaw ; ; ; ; ; ;register => 5552530146:7036361399 at 69.90.209.57/5552530146 register=>5552530146:boston!@#123...

...- Executing [105 at 105:2] Playback("SIP/voipvoip.com-0000000f", "hello-world") in new stack -- <SIP/voipvoip.com-0000000f> Playing 'hello-world.alaw' (language 'en') -- Executing [105 at 105:3] Dial("SIP/voipvoip.com-0000000f", "SIP/ sip3.voipvoip.com/17037171624") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/sip3.voipvoip.com/14445555514 [Apr 9 16:07:11] WARNING[994]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 4dd167154ea52bd26d63a95a56aa9526 at 192.168.1.10:5060 for seqno 102...

...use a IAX2 connection between version 1.2.14 and 1.6.1.0 but it gave an error - 1.2.14 End - Error Msg WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by 147.120.203.71: No authority found 1.2 END , IAX.conf [trunk14] type=friend host=147.120.203.71 secret=test123 context=sip,sip2,sip3 permit=0.0.0.0/0.0.0.0 1.6.1.0 End - Error Msg NOTICE[9854]: chan_iax2.c:8782 socket_process: Rejected connect attempt from 147.120.203.69, who was trying to reach '4567@' [trunk14] type=friend host=147.120.203.67 secret=test123 context=sip,sip2,sip3 keyrotate=off permit=0.0.0.0/0.0.0.0...

....98: No authority found -- Hungup 'IAX2/trunk14-9738' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/312-09f9a720' status is 'CHANUNAVAIL' [trunk14] type=friend host=147.120.203.98 auth=plaintext secret=Mah context=sip,sip2,sip3 ;keyrotate=off permit=0.0.0.0/0.0.0.0 1.6 EXTENSIONS.CONF [globals] TRUNKIAX14=IAX2/trunk10 at 147.120.203.98 [sip] ;exten => 4567,1,Dial(${TRUNKIAX14}/${EXTEN}|10|t) exten => 4567,1,Voicemail(${EXTEN},u) ~ 1.2 EXTENSIONS.CONF [Jun 1 05:20:31] NOTICE[9536]: chan_iax2.c:8782 socket_...

...e, even when all SRV > > > hosts have the same priority and weight. It should round > > > robin in this case. > > > > Agreed. > > This is how the polycom guy explain it. Lets say you do an srv lookup and > get: > > sip1.test.com > sip2.test.com > sip3.test.com > sip4.test.com > > The phone will try to register with sip1.test.com. If it is successful, > great. If not, continue to sip2.test.com, then sip3, sip4 and then back > again to sip1 and it will cycle untile it can find a server to register > with. Now lets say you are re...

...it's cache again, or there is a failure, even when all SRV > > hosts have the same priority and weight. It should round > > robin in this case. > > Agreed. This is how the polycom guy explain it. Lets say you do an srv lookup and get: sip1.test.com sip2.test.com sip3.test.com sip4.test.com The phone will try to register with sip1.test.com. If it is successful, great. If not, continue to sip2.test.com, then sip3, sip4 and then back again to sip1 and it will cycle untile it can find a server to register with. Now lets say you are registered to sip1.test...

> -----Original Message----- > From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] > Sent: Thursday, March 16, 2006 8:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] Feedback from VON expo! Info on > *HAandPolycomphone!! > > > > > > "Q: What are the plans for HA? > > That's BS. Last time I

Dear all, I'd like to know if there is a way for multiple asterisk servers to share a common SIP and/or IAX registry. The setup I imagine would be something like : - several asterisk servers called sip1.isp.com, sip2.isp.com, ... - a DNS alias sip.isp.com pointing to all the addresses (thus providing a round robin resolution on each server) - each SIP client would register with sip.isp.com

Hi all. I have 2 ip phones (Grandstream Budgetone): -budgetone1 -budgetone2 All two are connected to an Asterisk server. When I make a call from budgetone1 to budgetone2, I can speak with budgetone2 whith no problem. But when budgetone2 hangs up, budgetone1 does not play any tone (like busy tone). Budgetone1 seems to be still in conversation, but what conversation! Has anyone had a problem

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten => _89859716,1,Dial(SIP/202) [macro-sipmail] exten => s,1,Verbose(1,Extension ${ARG1}) ;line req to pick up ext if it's not reg. exten => s,n,Dial...

...NAT column is stating "N" I can call them and they can hear me fine, but I can't here them. I'm thinking this has to do with RTP, but not sure. In the router I have the following setup under "Virtual Server": SIP TCP/UDP 5060 IAX TCP/UDP 4569 KS1 UDP 5004 RTP1 UDP 5000 SIP3 UDP 5036 SIP4 UDP 2727 In the firewall section I've said to allow UDP on 9999-20001 to go to the asterisk server It looks like this in the firewall rules; Source *,* Dest *,192.168.x.x UDP,9999-20001 Also on those extensions that are coming from an external source I've added the externi...

...want to do features as belows. user ---> call ( from telco) --> asterisk ---> IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between user and SIP 2 (don't used call conference) SIP3 want to hear stream sound data of user and SIP2 conversation, can be press DTMF keys as: form example: *8401 ( 401 as username of SIP2). could you like to help me to implement that function. Best regards --------------------------------- Yahoo! Messenger with Voice. Make PC-to-Ph...

...I want to setting as belows. caller ---> call ( from telco) --> asterisk ---> IVR -- SIP 1. after that, SIP1 transfer to SIP2 (unattendant or attendant transfer). i want to SIP1 hear stream sound data of call conversation between caller and SIP 2 (don't used call conference) SIP3 want to hear stream sound data of caller and SIP2 conversation, can be press DTMF keys as: form example: *8401 ( 401 as username of SIP2). could you like to help me to setup that function. Best regards __________________________________________________ Do You Yahoo!? Tired of spam?...

...user1234,20,r) .. and if user1234 is registered, user1234 is dialed. But what about this multi-server environment? If the same extensions.conf line appears on all four asterisk servers, but the user is only registered to sip2.telco.com, how can the administrator make a Dial(SIP/user1234,20,r) on sip3.telco.comgo to the right user? Does a type of 'confederated sip registrations' system exist in Asterisk 1.4 ? Best regards, SJJD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070217/dd491302/a...

...alify=yes nat=yes dtmfmode=rfc2833 [221] type=friend username=221 secret=xxxxxxxxxxxxxxx host=dynamic call-limit=3 qualify=yes nat=yes dtmfmode=rfc2833 [222] type=friend username=222 secret=xxxxxxxxx host=dynamic call-limit=3 qualify=yes nat=yes dtmfmode=rfc2833 mailbox=422 vmexten=702 fromdomain=sip3.xxxxx.co.uk [223] type=friend username=223 secret=xxxxx host=dynamic call-limit=3 qualify=yes nat=yes dtmfmode=rfc2833 extensions.conf [default] include => blf exten => _2XX,1,SIPAddHeader("Alert-Info:<http://nohost>\;info=alert-internal\;x-line-id=0") exten => _2XX,n,DIA...

Ready to scream here.. 1. After 6 months with Asterisk I'm STILL trying to understand the difference between a SIP user, friend and peer. 2. Exactly what resource does Asterisk use to send MWI to registered phones? I thought it was astdb? 3. It looks like it isn't astdb. It looks like it will only send MWI to a phone if it shows up in 'sip show peers'. 4. WHY then does a reload