similar to: Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS

hello list i need your help please regarding an issue with snom300 and aastra6731i using asterisk 11.13.0 asterisk snom 300 8.7.3.25 astra 6731i 2.6.0.2019 i have configured the trunks like below 100 in snom300 200 in snom300 300 in aastra6731i 400 in x-lite the calls between x-lite and aastra ====ok inbound and outbound the calls between x-lite and snom300====> ok inbound and

Hi, This seems to be a fairly common question, but I have Googled for this quite a bit and looked at the Asterisk documentation/book and haven't been able to find an answer. My question is: Can I get my IP phone to select a different codec depending on the final destination of each call? I've got these things connected to my Asterisk box: - Snom 300 phone (supports g729 and

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from

You would need to give more information really. Your sip.conf file listing the entries for the phones especially which codecs are permitted. A copy of the 'asterisk -rvvv' console output when you make the call. On 27/03/15 17:05, Salaheddine Elharit wrote: > please no body has som with aastra can help me in this issue > > 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit >

Hi all, I face the subject strange behavior: calls arre dropped after 15 minutes on an asterisk 1.8.15.0. Only phones (SNOM300) connected to the Asterisk through OpenVPN seems to have the problem. From CDR, I see for 3 calls from this morning I'm aware of, that asterisk hangup after respectively 899s 894s 898s In logs I see WARNING[8213] chan_sip.c: Retransmission timeout reached on

Hi, I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months. Here are SNOM Phone and the firmware version; snom190-SIP - Version-Code: snom190-SIP 3.56m snom320-SIP - snom320 jffs2 v3.36 snom300-SIP - snom300-SIP 6.5.2 Asterisk version - Asterisk

Hi all, After the memory leaks (bug 1967 <https://bugzilla.mindrot.org/show_bug.cgi?id=1967>) I reported in bugzilla are fixed, I also applied melton(http://lcs.ios.ac.cn/~xuzb/melton.html) to detect the potential bugs in sshd (openssh-5.9p1). The url below is the index of bug reports that are checked as real bugs manually.

Ok here is another problem I have run into. I have a Sipura 2000 and I have been able to configure line 1 with only one small problem. But I can't get the line 2 working with asterisk. Here are samples of my sip.conf and extensions.conf. If I disable line 1 I can then get line 2 working. Is there a sample configuration for the Sipura to get both ports working with Asterisk. Sip.conf

Hello, One of our faculties have Microsoft's LCS and would like to connect it to our Asterisk system. the problem is that Asterisk talks SIP over UDP while LCS talks SIP over TCP with TLS. Anyone can recommend a gateway between these two protocols? Thanks! __Yehavi:

Hi List! does anyone played around with the LCS and Asterisk? Because the LCS is doing no RFC compliant SIP, i wonder if it can work. Google couldn't tell me. If someon heared about that, please let me know. The fact i figured out is that the Border Controler from Jasomi can be used as a gateway from MS-LCS-SIP to regular SIP. But that is not really handy and expensive too. Thank you

Search google with "sip pstn site:www.microsoft.com" You will find out how to configure LCS static routing to SIP Gateway, like Asterisk but you need patch Asterisk to support TCP. http://bugs.digium.com/view.php?id=4903 Step1: configure LCS 2005 to let sip uri: *@pstngw.domain to route to next hop: pstngw ip address Step2: patch your asterisk chan_sip.c to support TCP Step3: configure

Can anybody tell me if it makes a difference if domU''s have separate LVM volume groups? For instance, the Xen User Manual ( http://tx.downloads.xensource.com/downloads/docs/user/#SECTION03330000000000000000) says, when setting up a domU''s disks with LVM, to do a vgcreate vg /dev/sda10 Should each domU have it''s own volume group, or can all the domU''s share

https://bugzilla.mindrot.org/show_bug.cgi?id=1967 Bug #: 1967 Summary: Potential memory leak Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: unassigned-bugs at

Hi. I would like to thank Johannes, Piet and others for the lenny-cran AMD64 ports. I have a question about r-recommended from lenny-cran which I just installed: hotelling:~$ apt-cache policy r-recommended r-recommended: Installed: 2.9.0-1~lennycran.0 Candidate: 2.9.0-1~lennycran.0 Version table: 2.9.0-1 0 600 http://debian.lcs.mit.edu unstable/main Packages ***

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

I need a softphone - for usb phone devices - that I can alter (insert logo, menu, etc). Does somebody know such one? []s -- Abra?os Luis Claudio Mobile + 55 21 9215 2888 Mobile +55 15 9141 8402 Office +55 15 2102 5859 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/b8593cb1/attachment.htm

Hi! im trying tu run "microsoft office communicator 2005" and i cant resolve this: fixme:ntdll:NtConnectPort (0x1434f8,L"\\RPC Control\\epmapper",0x33ecd0,(nil),(nil),(nil),0x33ecf8,0x33ece0),stub! i google it all nigh long and i just cant find the way!!!. I need to connect to LCS 2005 because my company switch from Jabber to LCS. I tried pidgin and miranda-im+sip but didnt

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

Muchísimas gracias Carlos, de verdad que te agradezco la ayuda, pero no es lo que voy buscando. Quiero colocar en el eje de abscisas la secuencia temporal de los meses, es decir, agosto septiembre, octubre, etc? pero no las fechas de las toma de datos, sino que aparezca la marca de un mes, y la siguiente marca sea la del siguiente mes, etc?, y además que las muestras estén separadas de acuerdo con

Hi, everyone, Just wondering if someone had come up with a good way to do fuzzy searches if you use MySQL as your database (we tried switching to PostgreSQL, but that ended up adding even more problems). Ferret sounds great, but reading through the discussion it looks like we need to solve the problem of write conflicts. I just wrote a post in ruby-talk about using KirbyBase maybe to solve