search for: sip2

I get the following message on a Centos 5 system (really a Trixbox 2.4 build on Centos 5): Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled What does this mean? This message occurs about 30 times/sec for about 45 sec. Then my Bluetooth token starts up. Jan 14 00:12:28 sip2 kernel: hub 1-0:1.0: connect-debounce failed, port 1 disabled Jan 14 00:13:00 sip2 last message...

...etting the manager interface to behave properly; specifically the Originate event. If I create an originate event as below, the calling phone will auto-answer (as it's supposed to) but the receiving phone never rings. It will timeout at 20 seconds. Action: Originate Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 CallerID: John Doe <201> This is rather odd, as if I create a nearly identical call file in /var/spool/asterisk/outgoing (below) the receiving phone rings correctly. Channel: Local/201@from-sip2 Context: from-sip Extension: 154 Priority: 1 Calle...

Hello list, my case is as follows: SIP1--asterisk--SIP2. SIP2 is IVR type device. SIP1 and SIP2 both use g729. When SIP1 calls SIP2, it hears the IVR, and prompt the SIP1 to punch the keypad on the phone. As suggested by you, I need to configure the SIP1 with out band dtmf mode , what is about the sip.conf, should I specify the SIP1 with demfmode=rfc22...

Hi, I am trying to write dial plan for sip to auto answer (auto attend) the incoming call to the sip phone. - If i call from sip1 to sip2 then sip2 should automatically answer the call and play some sound file. I am trying to do this but as new to the asterisk dial plan configuration , so not able Todo this. help me if anyone already done this setup. Regards Upendra. -------------- next part -------------- An HTML attachment was s...

Hello list. When using SIP DNS SRV to define a production Asterisk server with high priority and a backup Asterisk server with a lower priority on this DNS-server, will this work as follow : - production server is reachable, so registration of the IP-phone goes to this server - production server is unreachable, so registration goes to the backup Asterisk server - production server is

Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten => s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate

...ing Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to 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 --&...

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can dial MeetMe (955) no problem but when 975 dials X-Lite,...

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 (thus ending on a random asterisk servers) - but after that, all the servers would be "aware" of the regist...

...getone's hangup. My sip.conf: [general] disallow=all allow=ulaw bindaddr=172.16.60.21 [sip1] callgroup=1 pickupgroup=1 type=friend secret=sip1 auth=md5 host=dynamic reinvite=no canreinvite=no callgroup=1 pickupgroup=1 dtmfmode=rfc2833 callerid="sip1" <101> context=telefonos [sip2] callgroup=1 pickupgroup=1 type=friend secret=sip2 auth=md5 host=dynamic reinvite=no canreinvite=no callgroup=1 pickupgroup=1 language=es dtmfmode=rfc2833 callerid="sip2" <102> context=telefonos extensions.conf: [globals] EXTEN106=Sip/sip1 EXTEN107=Sip/sip2 [telefonos] exten =&gt...

...seems like the right answer. What I want is for 2 SIP phones to use speex to talk to each other through 2 asterisk boxes (linked over IAX2) while only supporting ulaw on the asterisk boxes themselves. I think a diagram will help ;) SIP1 <--> *1 <--> IAX2 link <--> *2 <--> SIP2 I want any calls from SIP1 to SIP2 or SIP2 to SIP1 to be able to use speex OR ulaw (depending on network status) I want any calls to *1 or *2 to use ulaw only (VM and other features) since those should be over LAN anyway. I want the IAX2 link (which is over the internet) to transmit whatever the...

...meone 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 him domain provider2) traced this: Via: SIP/2.0 UDP sip1.provider2.de and it...

Another request make me test my t1 card, which has no quality problems, but all that I get is: [root@SIP2-MI zaptel-1.2.1]# ./zttest Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.987793% 99.98779...

...ed to 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....

Is it possible to split sip.conf into two files (sip1.conf sip2.conf)? I have an Audiocodes gateway with two FXO ports, and (according to info I received, and it appears to be correct) Asterisk find the peers based on their IP and not on their IP+PORT. Thus, Audiocodes with two FXO ports registered on the same devices (=> one single IP with different SIP p...

...0.203.98: 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 so...

...Username Refresh State Reg.Time 202.168.56.133:5060 61990xxxxxx 105 Registered Fri, 11 Apr 2008 15:15:58 sip.pennytel.com:5060 61289xxxxxx 105 Request Sent Thu, 10 Apr 2008 21:38:54 sip2.bbpglobal.com:5060 617000xxx 105 Request Sent Thu, 10 Apr 2008 20:43:20 sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip-register.conf': Found == Parsing '/etc/asterisk/sip-klav...

Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle

Hi All I 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 SI...

Hi All 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...