similar to: Help with asterisk and avaya SIP trunking

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23]

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

I have finally crashed Asterisk for the first time and I'm wondering if anyone has seen this. This is a configuration with SIP endpoints and an IAX2 channel to another Asterisk PBX. The main PBX dropped a core file after a SEGV (signal 11 ) with the following trace: #0 0x42079133 in strchr () from /lib/tls/libc.so.6 #1 0x41bb0f9c in _fini () from /usr/lib/asterisk/modules/chan_sip.so #2

Rommel Rodriguez Toiracrommelrt at nauta.cu On 18/11/2020 17:34, Rommel Rodriguez Toirac via samba wrote:>?? > In my network I have a samba 4.11.4 as Active Directory Domain Controller installed in CentOS 7 (gtmad.gtm.onat.gob.cu - 192.168.41.17). I have recently installed samba 4.13.2 in CentOS 8 (gtmad1.gtm.onat.gob.cu - 192.168.41.18) and following the wiki.samba.org guide I have joined

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

Hi all, I'm currently trying to bootstrap llvm-gcc-4.2 from the latest stable LLVM release 2.6 on Solaris SPARC. The compilation with a self-build gcc-4.2.4 goes pretty well, however, I hit a seemingly widely know error during llvm-gcc's bootstrap in getting an assertion during the compile of /data/LLVM/src/GCC/RELEASE_26/gcc/libgcc2.c: /data/LLVM/src/GCC/RELEASE_26/gcc/libgcc2.c: In

Hi Anton, thanks for the fast response. Ty->dump() prints "{double, double}" in the failing case (just before my introduced assert). HTH, and regards, Juergen On 10/30/09 16:19, Anton Korobeynikov wrote: > Hello, Juergen > >> With the full patch, I get the assertion of the 'unhandled REAL_TYPE!', >> so I wonder what needs to be done in order to get

Hello, Juergen > With the full patch, I get the assertion of the 'unhandled REAL_TYPE!', > so I wonder what needs to be done in order to get LLVM to produce > code for 'TREE_CODE(type) == REAL_TYPE' in the 'HandleArgument' function. > > > Does anyone have an idea? And why does this only happen on SPARC? What is the value of "Ty" in your case?

Can be done calls from each system? How about to capture data with Wireshark? I have experiences Asterisk with Panasonic with H323 without any problem. Mc GRATH Ricardo E-Mail mcgrathr at mail2web.com

Does anyone know if the Avaya 5610 supports SIP? I cannot even find the phone listed on the Avaya support site. Anyone have an example config? Thanks.

Hi List, I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i would like activate a "direct media path" for the RTP transit directly between the phone and the Asterisk. Now, - H323 Trunk is OK - RTP from the phone transit directly to Asterisk (activate "strictrtp=no" in rtp.conf, and "Allow Direct Media Path" option in Avaya Ipoffice) H323: Phone

Hi List, Ive got a bunch of Avaya IP9640 that we want to convert to SIP and then hook up to Asterisk so we can dump this overpriced Avaya system. Ive got ahold of the SIP firmware, but I cannot find anything on how to convert the phone itself to SIP, when I go into setup mode it wants a "command" which im guessing is the program code like the rest of the Avaya systems works. Has anyone

Has anyone successfully setup the Avaya IPOffice 500 with a sip trunk to Asterisk. If so can someone give some config examples? Thanks Rick -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070825/50bd7d63/attachment.htm

I just registered an Avaya 9620 set to my Astlinux system (0.47 - Asterisk 1.2.22), using Avaya SIP Firmware version 2.0.1.34. Set mailbox=ext at default in the sip.conf Found MWI worked immediately. Turned off as expected. Have Fun! Tom -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I'm new to the list, so I apologize in advance if I'm beating a dead horse by asking this, but I read somewhere that asterisk 1.4 has MWI working for Avaya and their rather troublesome SIP firmware. Can anyone verify this before I go changing phone systems around? Thanks Derek -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Has anyone put * in between an Avaya and Cisco system to connect two offices together? I was thinking about adding a SIP trunk on each side and getting Asterisk to pass calls between them. There is a leased line for bandwidth. Any tips/ideas on whether this is possible or dumb? Thanks.

I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware. I have successfully registered this phone to Asterisk BE as well as Asterisk 1.4.18 Almost everything is working well. Except for two issues. One of the problems is that the phone looses registration every now and then ad I have to re register. I have found a tip for this which I am testing if it will work, which

Hi All; In cisco IP Phone, I can assign for a features for the buttons at the Phone (one button to be for call pickup and one button to be for call forward and button to be for bridge and one button to be for another extension). How can I do the same thin in the Cisco IP Phones (or other IP Phones) while they are registering in Asterisk? Regards Bilal

I have as large customer that would like to repalce all their Avaya PBXs with a OpenSer/Asterisk solution. Now the plan is to replace their 12,000 Avaya desk sets with Polycoms. My question is has anyone successfully used with OpenSer and or Asterisk, if so I would like to talk with you about hiring you to build this in our lab envirnment. Bob G. bobg at techie.com- --

Has anyone tried using ooh323 in Asterisk to talk H.323 to an Avaya S8500 running Communications Manager 4 software? I have a potential customer who has such a system, and wants an Asterisk box to talk to it. Apparently they don't have SIP installed. I've successfully got ooh323 talking between two Asterisk boxes, so am just after some confidence regarding the Avaya, or any gotchas to be