similar to: SIP ACK // CSeq 0 => ZAP Channel hangup

Hello, I'm trying to follow the documentation of how to use gridBase, and I've reached the minimal code example below as my best effort. Can someone explain how to keep the column of boxplots on the same page as the rectangles (even though I've tried new = TRUE) ? Also, would it be hard / possible to match up the middle of each boxplot to the middle of each rectangle ?

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

Hi All, This is the first time I'm submitting a patch, and I hope it fixes more than it breaks. I'm putting it here, since John Todd mentioned a while ago about the heavy load Mark and crew have at Digium (doing such good work), so I thought all of us could test this first, and if ok submit for inclusion in CVS later if appropriate. This is an extension to work done earlier (sorry I

Just put a new phone in place with the latest firmware from Cisco. This is the first SPA501G we have with this firmware. In the Asterisk CLI we are now seeing the error message below about once every second. When we unplug the phone, the messages quit. NOTICE[15539]: chan_sip.c:9602 copy_header: No field 'CSeq' present to copy Thanks in advance for any assistance on this.

People: Did you have seen this message before? noc2pbx*CLI> -- Executing Goto("SIP/-081363a0", "ivr1|2000|1") in new stack -- Goto (ivr1,2000,1) -- Executing Answer("SIP/-081363a0", "") in new stack -- Executing Wait("SIP/-081363a0", "1") in new stack -- Executing BackGround("SIP/-081363a0",

as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

Hi all: I've no response for the last question with the same subject. Please excuse me for the extreme length of this mail, but I send 2 SIP traces. I have problem with * and 5300, when the incoming and outgoing call are routed thru the same SIP gateway (AS5300). Do I need to set an special things in sip.conf? First all, the * printout. Second, the 5300 trace. Thanks in advace, Gus

Hi Steve, What language is that please? We're using Perl and so far I haven't found an equivalent there. Thanks for your help. On Fri, 7 Jun 2019 at 12:10, Steve Edwards <asterisk.org at sedwards.com> wrote: > On Fri, 7 Jun 2019, David Cunningham wrote: > > > We have a need to record audio and allow the user to press any DTMF key > > to end the recording.