similar to: Basic question regarding re-INVITE

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-lists at wiztech.biz> wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >> chan_sip to quit ignoring re-invites from Vitelity. Our side ends

Hello, If you have canreinvite=no and a peer sends you a re-invite, what will Asterisk reply with? Thanks, -- David Cunningham, Voisonics http://voisonics.com/ US toll-free: +1 888 842 2720 UK: +44 (0) 20 3298 1642 Australia: +61 (0) 2 9037 2180

Hello, I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw] include = from-didww [from-didww] exten = 3130XXXXXXX,1,AGI("did.php") exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20) but when i run the rule it says chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension

Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

Hi List I have UAs registered with Asterisk and make outbound calls via ITSP1, everything is fine without re-INVITE. When people call 178, the actual number 112233445566 at ITSP1 network will be called. When UA or called telephone (112233445566) hang up, the call and associated channels are cleared. Sip.conf [general] canreinvite=no nat=no [ITSP1] type=peer host=A.B.C.D

Hi there noticed a strange thing in asterisk 1.6.2x 1.6.1x after one of the clients sends bye asterisk first sends invite to other side then after 200 ok it sends bye I am not sure but that could be some missconfiguration issue or a bug? so it's like this: side A sends bye to asterisk, asterisk responds with 200 OK to side A, then it sends INVITE to side B, expects 200 OK

Hello Folks. I'm wondering if anyone has any helpful hints. I recently upgraded to 1.4.11, and I'm having problems with pickup, both directed, and the pickup feature. My server is on the public internet, and all phones are behind a NAT router, somewhere else on the public internet. When a ringing phone is picked up by another phone, you have audio for a few seconds, then the call is

Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have

I've got a problem with a SIP Account I am trying to dial in with. The correct extension rings but when I pick up the call is not made and I get a busy signal. Dialing out works just fine - just calling this number doesn't seem to work. Any pointers? Thx Michael excerpt from sip.conf: [general] context=default port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0

Hi All, We have asterisk 11.23 running sip to vitelity and from there IAX trunks split off to where they need to go. We are having a problem getting chan_sip to quit ignoring re-invites from Vitelity. Our side ends up sending a reinvite which their side & they do not support us sending a reinvite. Ive tried: canreinvite=no which was supposedly replaced by: directmedia=no Can anyone shed

I am getting a log message of Failed to authenticate on INVITE to ... after months of a system working. I have changed nothing... What can cause this. I did some searching and tried setting in sip.conf (canreinvite to both yes and no - made no difference) by default I had no entry at all when this started happening. I am using sip phones, grandstream, cisco combination and all running sip.

I've setup Asterisk to work with Broadvoice for both incoming and outgoing calls. I can make outgoing calls, but when I try to receive an incoming call I see the following message on the console: [date] NOTICE[8661]: chan_sip.c:13178 handle_request_invite: Unable to create/find SIP channel for this INVITE It's registered with Broadvoice: Name/username Host Dyn

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new

Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. A1.A1.A1.A1 = IP-address Asterisk PBX AS.AS.AS.AS = IP-address Aastra PBX Aastra PBX makes a call

Hello, I have a issue between asterisk and windows based VoIP system (Client). Vendor SIP Server --> My asterisk --> Client Here is ethereal trace between asterisk and client. 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session description 2 0.042380 192.168.4.23 -> 192.168.3.222

sip debug shows that my * is trying to invite my spa and being told 404 Reliably Transmitting: OPTIONS sip:42.7.11.194 SIP/2.0 Via: SIP/2.0/UDP 128.9.0.39:5060;branch=z9hG4bK43efe1d7 From: "asterisk" <sip:asterisk@128.9.0.39>;tag=as39d40d19 To: <sip:42.7.11.194> Contact: <sip:asterisk@128.9.0.39> Call-ID:

Can anyone give an example of the difference between the following: canreinvite=no canreinvite=yes canreinvite=update Here is the problem: I have an 800 number sent to me via SIP from a national carrier. Asterisk gets the number and rings my desk phone. Asterisk has 2 NICs, one with public IP and private IP. My phone is on private IP, the inbound call is on public. My phone rings and I answer

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Hi All, We are using asterisk 16.5 and having an issue with the first re-invite after the call has been established. We can see the call gets up and you see in the logs the bridge type has changed and after that a re-invite is triggered. Is there any possibility to deactivate this kind of reInvite? We have some race conditions while have multiple asterisk in the call flow and the different