similar to: x-ast-orig-host - How is this IP taken ?

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

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

-----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

Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER

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, 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, 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 sorry for asking the same question again: here is my details: I've 50 exten in my sip and I'm using snom300 to my asterisk box this asterisk box is connected to another asterisk box using IAX trunk over 1MB full duplex line. I'm using g729 as the preffered codec. Can you please tell me how many calls can go at the same time without causing the any type of problem. thanks arun

please no body has som with aastra can help me in this issue 2015-03-26 11:02 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>: > 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

Hi, I've had a Snom 300 connected to my Asterisk box at home for 12 months or so now. Recently it lost all its settings and I had to reconfigure it via the built in website. For a few weeks it was fine. Couple of days ago it lost its settings again. I logged in to its web server and thought I would upgrade the firmware. It seems to be running an old version: Phone Type: snom300-SIP

Hello, I am trying to add a sip-trunk to my Asterisk 1.4.15/Elastix 0.9.2 server. The system is in production with local extensions, a zap trunk and a working sip trunk with sipgate.de. My asterisk server is behind a NAT/Firewall, anyhow it registers and works well with sipgate.de on incoming and outgoing calls. I aquired an account with a reseller net-voz.com: I did some testing with the

Regid, You suggested we remove nut_version.h from the .orig.tar.gz for NUT: <http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=613643> The original intent was that nut_version.h would be generated from "make dist" (or "make distcheck*") when the official nut-X.Y.Z.tar.gz tarball is created. At that point, it is safe to assume that there is no longer any local version

thank you for your response below the asterisk -vvvr Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [0176XXXXXX at from-internal:1] Macro("SIP/300-00000192", "user-callerid,LIMIT,EXTERNAL,") in new stack -- Executing [s at macro-user-callerid:1] Set("SIP/300-00000192", "TOUCH_MONITOR=1427481319.470") in new stack --

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here: http://pastebin.com/XqZG1m5X I am connection over TLS / SRTP on port 5063. When

Hi...rsync working fine for my fedora 5 box to fedora 9 box. Now i want to take back of my folder /home/rajiv to a windows 98 box...Shared a folder in win98 system for full access but do not know how to take backup...tried with the following but got error message [root@myserver ~]# rsync -aPrv /data/stock/ //192.168.1.75/dir sending incremental file list rsync: mkdir

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no

Must be able to pass Caller ID number. Email me with your terms.

Hi, The bounce message generated by the reject extension has what looks like a hard coded message prefix that comes before the configurable reason text: "Your message to <username> was automatically rejected:" In some cases, the <username> is NOT the original-to address, which can cause confusion to the sender or expose private aliasing data that some people might want to

I've seen this happen three times in the wild now.  I've been trying to isolate the source of the issue, but so far it seems like there's not enough debug output to know why this occurs. Long story short: - Start Asterisk - PJSIP Handles receiving INVITE from ITSP via WAN (Asterisk is behind NAT).  SIP is handled correctly, Asterisk responds OK with RTP media address of