search for: 200ok

...y asterisk but the 200 OK message keeps routing between asterisk and opensips and then asterisk times out with ?*Maximum retries exceeded on transmission*? error. Scenario: 1) User ----7000 at sipproxy.com--------? Opensips --------7000 at asterisk.com-----?Asterisk 2) Asterisk ------200ok-----? Opensips---200ok---?User 3) Asterisk waits for extension input and user presses 7010 4) Asterisk------INVITE 7010 at sipproxy.com--------->Opensips<https://remote.novanet.net/owa/redir.aspx?C=31661696fae74c3a94f26b78ea106eae&URL=mailto%3a7010%40sipproxy.com---------%253e...

Hi, Can someone share how can I configure asterisk to get P-Associated-Uri header in 200 Ok to the REGISTER. Thanks, Amit -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120223/633da268/attachment.htm>

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE w/T.38-> ------INVITE w/ T.38--> <-----488--------------...

...B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood you Josh, there is no way to prohibit this kind of reInvite? It is not about route Optimization just for some more options for the A Party. BR Jöran On Thu, Aug 15, 2019 at 4:07 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Thu, Aug 15, 2019, at 8:23...

...t few weeks we had several crashes on live asterisks running versions 12.2.0rc1 / 12.6.1 with PJPROJECT versions 2.1.0 / 2.2.1. We opened a ticket - ASTERISK-24471. After investigating the issue I can say that the scenario is a CANCEL being received while handling ANSWER and before generating the 200OK response. Looking at the core file we see that the problem is in - pjsip/src/pjsip/sip_transaction.c line 3158 : PJ_ASSERT_RETURN(event->type == PJSIP_EVENT_TX_MSG && event->body.tx_msg.tdata == tsx->last_tx, PJ_EINVALIDOP)...

...ASTERISK-IP:6011 It looks like the SBC discards this Contact header as invalid and returning the 200 OK without contact to the registering client, indicating an unsuccessful registration. All other clients I have tried registering directly to asterisk seem to ignore this port and just accept the 200OK. Only our SBC causes this problem. I have attempted rewrite_contact yes and no, both with the same result. So from my point of view, Asterisk is putting the 'remote' port instead of it's own SIP port into the Contact Header. Can anyone confirm this is misbehavior be pjsip? Could this...

We are using an MGCP configuration. There seems to be some incompatibilities between our Netergy T2 VOIP chip and Asterisk. This is how our Vendor sees it: Here's what I see. 1. The first call is initiated. (CRCX) The interesting thing here is that the CA (Call Agent) tells us to go directly into sendrecv mode which means that we start streaming audio immediately. All other CAs that

Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows RTP to use different IP as SIP ip. Is there any way to configure it? GUI or CLI? or , will we support it in future? Thanks. -- Rgds, -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin at gmail.com MSN: hansyin at hotmail.com Skype: hans_yin_vancouver

...oks like the SBC discards this Contact header as invalid and > returning the 200 OK without contact to the registering client, > indicating an unsuccessful registration. > > All other clients I have tried registering directly to asterisk seem to > ignore this port and just accept the 200OK. Only our SBC causes this > problem. > > I have attempted rewrite_contact yes and no, both with the same result. > > So from my point of view, Asterisk is putting the 'remote' port instead > of it's own SIP port into the Contact Header. > > Can anyone confirm thi...

...2.0.0 working nicely together with no apparent errors. I connect to the URL with winamp and it works get. Problems arise when I try to connect with any other computer other than my own. This includes other computers on my LAN and off it aswell. In the winamp window they get the message: [HTTP/1.0 200OK] http://www.feelthevibe . . . Any suggestions as to why my computer is fine and everyone else's doesn't work. We're all using the same version of winamp (5.05) Any help would be much appreciated. Rob

Hi guys, My asterisk didn't send ACK for 200 ok message just for one specific extension. The ATA used by this extension is used by other extensions, with same firmware version. Looking in wireshark, I saw that ATA sent 200ok and asterisk didn't confirm it with ACK. The ATA did this during 20s, after this, asterisk hangup the call. This issue happen only one asterisk start a call. I'm using the latest version of asterisk. What can be the source of this problem, and how can I debug this problem in asterisk ? T...

...(Closes issue ASTERISK-22197. Reported by Dalius M.) * --- chan_sip: Do not increment the SDP version between 183 and 200 responses. (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok (Closes issue ASTERISK-22428. Reported by Ben Smithurst) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.25.0 Thank you for your continued support of Asterisk!

...responses. (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls (Closes issue ASTERISK-22005. Reported by Torrey Searle) * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok (Closes issue ASTERISK-22428. Reported by Ben Smithurst) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0 Thank you for your continued support of Asterisk!

Hello Everyone, We are seeing many instances where we receive 200OK from the vendors however, asterisk still keeps ringing. Is there anyway to stop this from happening? I remember reading something about early media however this seems to be a case of late media? Kind Regards, Nick from Toronto. -------------- next part -------------- An HTML attachment was scrubb...

...PJSIP. When receving an INVITE with FROM URI that has a port number, the Asterisk removes the port from URI on consecutive Responses / Requests. This causes an issue with one of our SIP servers (it doesn't recognize the response / request). Below you can see an incoming INVITE and the outgoing 200OK response. I have highlighted the issue in Yellow. Does anyone know of a solution / workaround for this issue? <--- Received SIP request (648 bytes) from UDP:172.16.60.160:5061 ---> INVITE sip:039988120F at 172.16.60.160:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060;branch=z9hG4bK-29...

...(Closes issue ASTERISK-22197. Reported by Dalius M.) * --- chan_sip: Do not increment the SDP version between 183 and 200 responses. (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok (Closes issue ASTERISK-22428. Reported by Ben Smithurst) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.25.0 Thank you for your continued support of Asterisk!

...responses. (Closes issue ASTERISK-21204. Reported by NITESH BANSAL) * --- chan_sip: Allow a sip peer to accept both AVP and AVPF calls (Closes issue ASTERISK-22005. Reported by Torrey Searle) * --- chan_sip: Fix Realtime Peer Update Problem When Un-registering And Expires Header In 200ok (Closes issue ASTERISK-22428. Reported by Ben Smithurst) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.7.0 Thank you for your continued support of Asterisk!

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

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

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,