search for: sdolloff

...anslator frame not getting freed. This issue prevented G.729 licenses from being freed up. (Closes issue #17630. Reported by manvirr. Patched by dvossel) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Q931 - Sending PROGRESS after sending ALERTING is a protocol error. (Closes issue #17874. Reported, patched by nic_bellamy) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLo...

...y leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup when notifycid is enabled. A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should on...

...sts with multiple contact headers. Patched by jpeeler. * app_followme: Don't create a Local channel if the target extension does not exist. (Closes issue #18126. Reported, patched by junky) * Revert code that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, te...

...issue #18384. Reported, patched, tested by bjm, tilghman) * app_followme: Don't create a Local channel if the target extension does not exist. (Closes issue #18126. Reported, patched by junky) * Revert code that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, te...

Are there any known issues with call-waiting caller-id for SIP? Caller-ID on the first call works fine, but when the second call comes in, I hear the interrupt tone, but the caller-id doesn't display anything. I have tried this with the Cisco ATA and the SPA-2000. I have also tried two different phones to verify that it wasn't something specific to the phone. Thanks, Stephen

...76591212@_sip.test.net>' failed for '209.242.0.1' How can I get more information on what is causing the failure? This same user authenticates fine most of the time, but I still get these types of messages much too frequently. Stephen Dolloff DLS Internet Services 847-854-4799 x256 sdolloff@noc.dls.net

Hello, One of the Sipura 2k's I'm using has a dialtone that occasionally fades to static when the user picks up the line. Are there any settings that I can check that would affect this? Regards, Christopher

...anslator frame not getting freed. This issue prevented G.729 licenses from being freed up. (Closes issue #17630. Reported by manvirr. Patched by dvossel) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Q931 - Sending PROGRESS after sending ALERTING is a protocol error. (Closes issue #17874. Reported, patched by nic_bellamy) For a full list of changes in the current release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLo...

...y leak. (Closes issue #17774. Reported, patched by kkm) * Add Danish support to say.conf.sample (Closes issue #17836. Reported, patched by RoadKill) * Ensure SSRC is changed when media source is changed to resolve audio delay. (Closes issue #17404. Reported, tested by sdolloff. Patched by jpeeler) * Only do magic pickup when notifycid is enabled. A new way of doing BLF pickup was introduced into 1.6.2. This feature adds a call-id value into the XML of a SIP_NOTIFY message sent to alert a subscriber that a device is ringing. This option should on...

...sts with multiple contact headers. Patched by jpeeler. * app_followme: Don't create a Local channel if the target extension does not exist. (Closes issue #18126. Reported, patched by junky) * Revert code that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, te...

...issue #18384. Reported, patched, tested by bjm, tilghman) * app_followme: Don't create a Local channel if the target extension does not exist. (Closes issue #18126. Reported, patched by junky) * Revert code that changed SSRC for DTMF. (Closes issue #17404, #18189, #18352. Reported by sdolloff, marcbou. rsw686. Tested by cmbaker82) * Resolve issue where REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. (Closes issue #18051. Reported by eeman. Patched, te...

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify

Can someone please explain what I am doing wrong here? I only want the extensions listed in long-users to be able to access the longdistance context. If I do this, I get a congestion tone no matter what I dial. If I add a [default] context and include => longdistance, then the local callers can call the long distance number fine, which is not what I want, but I still want long-users to be

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Has anyone succesfully integrated * with a cisco voice gateway ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031214/3b1ba7b3/attachment.htm

Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video attachment while using any video phone. This does work in my 1.8.23.1 installation. The file is skipped with the ast_streamfile error (and moved to OLD), and the prompts following that message display the ast_best_codec error. [Apr 7 16:05:50] WARNING[17497][C-00006fdd]: file.c:1017 ast_streamfile: Unable to open

I am having issues with Privacy Manager and Zapateller. If I set callerid="" on a sip user zapateller sends the tones If I set callerid="Anonymous" <8475551212> zapateller doesn't send the tones If I call from a phone after dialing *67 zapateller doesn't send the tones In the last 2 cases, the display on the phone shows -Blocked Call- PrivacyManager always gives

Greetings, I have a couple of questions and figured I would put them all in one message to not spam the list as much as possible. I have searched voip-info, google and the list archives for all of these questions. If I have missed the correct response, please accept my apologies. I have been stuck on these for a long time and I am really hoping that the other users out there will be able to

I really hope that someone can help me with this one. DTMF tones are not working for certain places that I call, specifically 1-800-882-8880 which is the AA advantage line. It works for almost everyplace else. If I bypass asterisk, the call works fine. Network looks like: <SPA-2000> --SIP-- ASTERISK --SIP-- <AS5350> --PRI-- PSTN sip.conf entries [VGW01] (this is the AS5350)

I would like to move voicemail to a dedicated server but I can't figure out how to make the MWI work if the ATA doesn't register to the voicemail server. The main reason for this is redundancy. I have two SIP registrars running and in the case of a failure from the primary, both the gateways and the ATAs switch over to the secondary, but since the voicemail is on the primary, it also