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On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> wrote: > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" document, it > occurred to me that the desired behavior should actually happen > automatically, just due to the codec negotiation logic, but it looks like > asterisk doesn't actu...

...or clarifying it. Any idea on the timeframe of implementation? *Michael Ulitskiy* Ace Innovative Networks, Inc. Main/SMS: 212-868-2366 Direct/SMS: 212-812-1203 https://www.aceinnovative.com On 7/6/23 12:47, Joshua C. Colp wrote: > On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com> > wrote: > > Hello, > > After I have re-read the "PJSIP Advanced Codec negotiation" > document, it occurred to me that the desired behavior should > actually happen automatically, just due to the codec negotiation > logic, but it looks...

Hello, I've started to play with PJSIP and got stuck at the following problem. I need to retrieve SIP Call-ID associated with PJSIP channel. For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER seem to be unable to read headers for outbound channel. Here's what I do:

Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the

Hi, Has anybody managed to get callerid properly set on a call from local to asterisk SIP endpoint through h323-pstn gateway to a regular phone. I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it. When I place a call to pstn I'm not receiving 12125551234 as the clid, but a number assigned to PRI channel by phone company. It worked with chan_oh323, but there were other

Hi, I'm wondering if there's a way within a dialplan or AGI to find out if an extension (SIP client) is already in use and the person is already on the phone? By default the channel is assumed available and callwaiting tone is transmitted to the called extension. AFAIK there's no way to turn off callwaiting from within the dialplan. I need to avoid the callwaiting behavior in some

...ineer for what ever amount of time or project we work out. This can be asterisk related such as IVR menus et al or music related, tracking, overdubs, mixing, mastering. The studio is located in Brooklyn, near the Brooklyn Childrens Museum. Please reply off list if your interested. bassmint sat acedsl not com J.P. Edmund "If you think it's not a game, you've already lost"

Hello, After I have re-read the "PJSIP Advanced Codec negotiation" document, it occurred to me that the desired behavior should actually happen automatically, just due to the codec negotiation logic, but it looks like asterisk doesn't actually follow the described logic which is likely a bug. Can you please follow with me through a simple sip call and see if I'm missing

I have been searching all over for the answer on all sources online and have come to the conclusion that it must be rudimentary or I am asking the wrong question. I cannot figure out how to configure the box to set the "from" address to a correct domain, as my outgoing isp will not pass mail from root@asterisk1.local, as I expect it wouldn't. Any help is appreciated, even just

Hi, Say I want to display the subversion release number in the footer of each page in order to track what version of a site I''m looking at. I know subversion has a substitution keyword (LastChangedRevision) that inserts the last known revision in which that file .changed. So, if I stick $LastChangedRevision$ into views/layouts/application.rhtml, it''ll show the last time that

...info/samba Return-Path: <freyes@inch.com> Delivered-To: samba@samba.org Received: from koza.acecape.com (koza2.acecape.com [66.9.36.222]) by lists.samba.org (Postfix) with ESMTP id DD7A245A1 for <samba@samba.org>; Tue, 24 Jul 2001 21:01:29 -0700 (PDT) Received: from miguel (p65-147.acedsl.com [66.114.65.147]) by koza.acecape.com (8.10.1/8.9.3) with SMTP id f6P46Y909095; Wed, 25 Jul 2001 00:06:35 -0400 (EDT) Message-Id: <200107250406.f6P46Y909095@koza.acecape.com> From: "Francisco Reyes" <freyes@inch.com> To: "Eagen, Dave" <eagen@BIPERF.com>,...

Hi, Does anyone know if asterisk can handle 3xx SIP responces? I'm trying make it work with redirect server and it looks like asterisk isn't going to send another invite, but treats "302 Moved Temporarily" message as "Everyone is busy". Thanks. Michael

Hi, I'm wondering if there are any plans on adding secondary gatekeeper support to asterisk h323 channel drivers. Also I've noticed that chan_h323 is crashing asterisk at startup if primary gatekeeper is not available. Wouldn't it be a more correct behavior if it doesn't crashing but continue registration attempts in the background? Didn't test it with chan_oh323. Thank you.

Hi, I'm sorry to ask this question, but I thought I'd rather ask it here before messing up with cisco. Is anybody running cisco 7960 in redundant configuration? I mean I want the phone to be registered with both primary and backup proxy (asterisks) so that service continues to work in case of primary proxy failure. I've set in SIPDefault.cnt: proxy1_address: 192.168.1.10

Hi, I guess I need some help with management interface. I would like to watch calls through the management interface, but I don't know how to identify which call an event belongs to or in other words how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/12125551111@pbx1 callerid: 12125551111 MaxRetries: 1

Hi, I have the following topology: PSTN/H323 gateway->GNUGK->chan_h323/chan_sip->SIP EP Mostly everything works fine except chan_h323 is not passing audio from PSTN before the call is answered and as a result users can't hear PSTN announcements (like "the number is not in service") that's played on unanswered call. All they hear is just continuous ringtone as though

Hello, I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on outbound (called) channel and has no affect on calling channel. My experiments and function documentation (which says “Media and codec offerings to be set on an outbound SIP

Well, I'm trying to migrate to chan_pjsip so that I don't have to do that. It's so surprising that the issue so seemingly obvious and trivial hasn't been addressed yet that I wanted to query the collective wisdom of this list to verify my observations. Thanks for github pointer. Michael On 7/5/23 16:46, asterisk at phreaknet.org wrote: > On 7/5/2023 4:19 PM, Michael

Hi, I don't seem to be able to get music on hold to play normally. The sound gets often interrupted with a few seconds of silence then starts playing again. I'm using mpg123-0.59r and tried mp3 files with different sample rates with no luck. If that matters, endpoints are SIP ata186, SIP Cisco 7960 and H.323 (over chan_h323) Quintum Tenor. Sometimes it may play fine for a few minutes

Hi, I have asterisk functioning as SIP to H.323 gateway for local SIP endpoints and I have H.323 to PSTN gateway (Lucent MAX TNT) connecting my LAN VOIP to PSTN via PRI. Everything works fine with one exception. I seem to be unable to figure out why I cannot hear PSTN intercepted announcement ("number is not in service" etc.) when I'm calling a disconnected number through