search for: sip1

Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is forced for the entire call. +--------+ +--------+ +--------+ | gs | <-----> | sip0 | <-----> | sip1 | +--...

...552 secs ago) SIP/Peggy (ringinuse disabled) (dynamic) (In use) has taken 1 calls (last was 822 secs ago) Callers: *1. DAHDI/i1/9705541916-1507 (wait: 4:32, prio: 0)* *core show channels concise * SIP/KWakmn-0000181a!LocalSets!sales!1!Ringing!AppQueue!(Outgoing Line)!214!!!3!1!(None)!sip1-1367428777.13318 DAHDI/i1/7812693000-1508!queues!sales!21!Up!Queue!sales,tc,,,,,,sub-QueueConnected!7812693000!!!3!277!SIP/Peggy-0000180c!sip1-1367428501.13296 SIP/Erin-00001819!LocalSets!sales!1!Ringing!AppQueue!(Outgoing Line)!233!!!3!8!(None)!sip1-1367428769.13317 DAHDI/49-1!queues!sales!21!Up!...

...all we dont know about. Cseq 42686 Cmd SIP/2.0 That itself would not be a problem, but my provider is complaining about lots of faulty registrations. I ran ngrep on that traffic. This is ngrep output after fresh start of asterisk: U 200.100.100.123:5060 -> 195.195.12.223:5060 REGISTER sip:sip1.provider.com SIP/2.0..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK6cb7aba7;rport..From: <sip:4221917293125@sip1.gtsne xtra.sk>;tag=as04cc5746..To: <sip:4221917293125@sip1.provider.com>..Call-ID: 0c74fd93144a92174f3a032d4aeca8dc@200.100.100.123..CSeq: 102 REGIST ER..User-Agen...

...== ast_get_enum() profiling: FAIL, 8.7.2.2.6.6.2.0.0.8.1.e164.org, 405 ms -- Executing [18002662278 at outbound:3] Set("SIP/aastra-sip1-0c004d98", "ARRAY(i,id)=1,0") in new stack -- Executing [18002662278 at outbound:4] Set("SIP/aastra-sip1-0c004d98", "max=3") in new stack -- Executing [18002662278 at outbound:5] While("SIP/aastra-sip1-0c004d98", "1") in new stack...

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 wit...

Without using 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...

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

Sipdiscount has replaced their asterisk servers for another thing. Then, no more iax. Ok, but I can't make calls using sip also... I'm getting a "forbidden" error when using sip1.sipdiscount.com. Anybody got it working? -- Alejandro Vargas

I am trying to move from IAX2 to SIP for voipbuster, moving at the same time to sip1.voipbuster.com. When I try calling out, I see that there is SIP exchange, and in many cases also RTP data being exchanged. Hover in a very large number of attempts the connection is not established. Half of the time there is no RTP, the rest of the time there *is* RTP data flowing in two ways, bu...

...> possible in recent versions of Asterisk either with chan_sip or pj_sip? > > PJSIP does not currently have functionality to allow such a thing. I > believe in chan_sip there have been no changes to remove it. > My DP looks like this: Exten => aaa,1,Dial(SIP/USERNAME:PASSWORD at sip1.myproxy.net/18005551212) and from the logs I get: oice1*CLI> console dial aaa at from-external -- Executing [aaa at from-external:1] Dial("Console/default", "SIP/ USERNAME:PASSWORD at sip1.myproxy.net/18005551212") in new stack [2019-07-09 08:40:54] NOTICE[27159][C-0001...

...------------------------------------------------------- Hi, I'm having trouble explicitly forcing a codec between sip devices. Am I missing something or is this not really possible? I have a grandstream registering to asterisk, named sip0. Sip0 registers, via sip, to another asterisk box, sip1. When I place a call from the grandstream, it will travel through sip0 to sip1, where it is then placed to the PSTN. Nothing can reinvite, this path is forced for the entire call. +--------+ +--------+ +--------+ | gs | <-----> | sip0 | <-----> | sip1 | +--...

...x110736c .Transaction created for Request msg REGISTER/cseq=36181 (tdta0x721d90) [Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c Sending Request msg REGISTER/cseq=36181 (tdta0x721d90) in state Null [Dec 22 19:24:24] DEBUG[25247] pjsip: sip_resolve.c .Starting async DNS A query: target=sip1.easybell.de, transport=Unspecified, port=5060 [Dec 22 19:24:24] DEBUG[25247] pjsip: resolver.c .Transmitting 34 bytes to NS 0 (192.168.178.1:53): DNS A query for sip1.easybell.de: Success [Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .State changed from Null to Calling, event=TX_...

...setup a SIP channel between two Asterisk box, and use Manager API to generate some calls. It's working quite fine, except this message (on the caller-side) : Jan 10 18:18:09 WARNING[25046]: chan_sip.c:6805 handle_response: Forbidden - wrong password on authentication for INVITE to '"sip1" <sip:asterisk@192.168.1.200>;tag=as77e9ebbb' But the call is going through and I can answer on the other side. Could someone give me some help about the meaning of this warning ? What am I doing wrong ? Thanks. Here my sip.conf : ***BOX1*** [sip1] type=friend context=answer-...

...39;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 regis...

...until it refreshes > > 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 y...

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

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 attachme...

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

...from an analog phone on a Zap channel using a SIP provider works fine as well. HOWEVER, when dialing out using a SIP provider (both Nikotel and iConnect) Asterisk cannot bridge the two legs of the call and all I get is silence. here is what the console shows: -- Executing Dial("SIP/Sip1-1862", "SIP/442071231234@nikotel|60|r") in new stack -- Called 442071231234@nikotel -- SIP/nikotel-4815 is ringing -- SIP/nikotel-4815 answered SIP/Sip1-1862 -- Attempting native bridge of SIP/Sip1-1862 and SIP/nikotel-4815 == Spawn extension (internal, 004420...