similar to: Patch to fix some segfaults in Asterisk

On 06.11.2020 13:32, James Henderson wrote: > Hi Alexey, > > On Thu, 5 Nov 2020 at 21:02, Alexey Lapshin <avl.lapshin at gmail.com > <mailto:avl.lapshin at gmail.com>> wrote: > > Hi James, > > On 05.11.2020 17:59, James Henderson wrote: >> (Resending with history trimmed to avoid it getting stuck in >> moderator queue). >>

The file "rtp.c" currently contains the following hack for special-case handling of some Cisco-specific protocol: } else if (payloadtype == 121) { /* CISCO proprietary DTMF bridge */ f = process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen); As I noted in my earlier message, I'm

hi! anyone try siemens optipoint 400 economy SIP phone with * ? -- http://www.siemens.com/Daten/siecom/HQ/ICN/Internet/Enterprise_Networks/WORKAREA/skuch_c/templatedata/English/document/binary/a31002-h1000-a250-2-7629.pdf Thomas

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

http://bugzilla.mindrot.org/show_bug.cgi?id=178 Summary: Content of /etc/nologin isn't shown to users, fix triggers probably AIX bug Product: Portable OpenSSH Version: 3.1p1 Platform: PPC OS/Version: AIX Status: NEW Severity: normal Priority: P2 Component: sshd AssignedTo:

Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Address 0xfffffffe out of bounds why and how to solve.MyConfbridgeCount(conferencenumber,variablename )return total number of user in conference given by conferencenumber otherwise zero.At runtime using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call function count_exec(struct ast_channel *chan, const char *data).But at compile time char * data cause core dumped.

Am 27.06.2000 19:58:44 schrieb urban: > On Tue, 27 Jun 2000 klaus-georg.adams@rwg.de wrote: > > > > > Hi Andrew, > > your patch from 2.2.15 to 2.2.16, backing out the older protocol levels breaks > > reading from an OS/2 LAN Server. > > The appended patch fixes things for me (against 2.2.16). > > This backs out a lot of desired changes. For example I think

Hello all, I'm looking for some help to try to understand why my CPE doesn't work good with Asterisk in MGCP. Here is what I want to do : - Register a TECOM AH4021 on Asterisk in MGCP with the following profile in mgcp.Conf : [general] port = 2727 bindaddr = 10.95.20.1 disallow=all allow=g729 allow=alaw 020202020202] context=mgcp host=dynamic canreinvite=no dtmfmode=rfc2833 nat=yes

The Asterisk Development Team has announced the release of Asterisk 1.8.14.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 1.8.14.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

I have found some references to the "hint" (or HINT?) variable and method in the source code, but quite a bit of Google-ing has not turned up any extensive answers as to some real-life examples of how to use this perhaps very useful tool. I understand the point of the tool, but I need to get some actual configs to look at before I think I'll figure it out. Even if my

Hi there, I'll need some help with this: Trying to establish an IAX2 link between two servers works in one direction (SIP client with ulaw), but not in the other (SIP client with GSM). The client used for this is X-Lite behind NAT while both servers have a public IP (I playback an anouncement before trying to connect to the second *). Error on the originating * server:

I have 1 IP phone (Swissvoice IP10S) and 1 POTS phone. When I dial the number for the IP phone off the POTS phone, the IP phone rings. But when I pick up the handset on the IP phone, I get a busy signal and this message on *: Aug 23 09:38:57 NOTICE[1142106560]: chan_mgcp.c:2243 handle_response: Terminating on result 502 from svip10@00059002042b-1 Here is the entire session. svip10 is the 1 and

Dear HO SIN, many thanks for "L" solution you give me... I have also libtiff troubles with RH9, how did you resolve them ? R2 protocol I use work only with AB bits(CDbits ares fixed) and it run perfectely with the testcall.c, i'll try with chan_unicall and i'll give you the result... *** GOOD luck Steve for updating the patch of last asterisk *** thank you for your help

Just an FYI to all out there that are upgrading after this weekend's run of CVS updates that are in now... MAKE SURE YOU DO "make clean". If you don't and asterisk acts funny this is why. Anytime any struct like ast_channel (which was changed over the weekend) and you don't make clean you'll end up with an asterisk box that acts retarded. So please before reporting a

Hello! Looks like h323 compiling is FAQ, but I didn't found an answer... The same problem with 0.6.5 and 0.7.1: gcc -Wall -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -I/var/local/files/asterisk-1.0.5/include -I../wrapper -g -c -o chan_oh323.o chan_oh323.c chan_oh323.c: In function `oh323_call': chan_oh323.c:1454: error:

I've just upgraded to the latest CVS head, and my outbound calls stopped working. I traced it back to the line exten => s,9,CallingPres(${ARG2}) It seems as if this application is now missing. I tracked back the changes and found in 1.415 of chan_zap.c the code was removed because it was "duplicated". However, it does not exist anywhere ! Am I being stupid, missed