similar to: How to pass through supported 100rel

Hi I want to know how to solve below an error case. Uac cant's change username of from and digest header. I tried to put aaaa at 192.168.0.1 on username of sip.conf.but same error returned. [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1 [Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479

Hi I am trying out Asterisk 12 between two servers in order to test the 100rel (PRACK) capability. When I set the called server to send Ringing (180) and wait before answering the call it is not sending 100rel Require in the Ringing message (non provisional response) therefore the calling server does not send a PRACK. I have the same config at both ends and if I change the 100rel field in the

Hi, I've used various patches with asterisk 1.4 to have support for call pickup and notification with good results. Now I'm trying vanilla 1.6.2 with its official support for "dialog-info +xml" notifications with no success. This is what i'm doing: - Phone A has a key configured as type "extension" pointing to Phone B. - In sip.conf I added

I am having a problem with one of my callers who is using either g729 or alaw.  I can do alaw but not g729 so asterisk should negotiate alaw right?  In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up.  Why? Last minute thought: Is it possible that

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

hi, let me explain in detail, what i have configured and what is happening now: 1st router w724v (Deutsche Telekom AG): - port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router) 2nd router Bintec RS353j): - configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports) other direction is totally open. I

-----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

OK, it stopped working. It turns out the transport and endpoints in PJSIP are ok. I can send an invite from my unregistered snom phone and I can see some activity in the CLI. However, when I dial from my snom to Kamailio and have it pass the message to asterisk, PJSIP seems to ignore the sip messages even though they are there. Is there something wrong in the invite that I'm missing? U

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =

Hello All, I've spent a day trying to get a Polycom IP500 wokring with my Asterisk box. I have several others that are working fine, but this one is getting by me. Can someone on-list tell from the following SIP debug what I've missed? Sip read: INVITE sip:2000@192.168.1.30:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK51f01c8152912F0E From: "2004"

I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP

On Mon, Jun 5, 2017, at 04:26 PM, Michael Maier wrote: > On 06/05/2017 at 06:29 PM, Joshua Colp wrote: > > On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > >> > >> Do you have any idea where to start to look at? Adding additional output > >> in the source code? Which functions could be interesting? I may add own > >> debug code to see why things

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The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Is it possible to use the instant messaging feature of Polycom phones in Asterisk? At the moment I'm seeing this in the SIP messaging when I try to send one from a Polycom 450. <--- SIP read from UDP:<CENSORED POLYCOM IP>:5060 ---> INVITE sip:0100@<CENSORED>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <CENSORED POLYCOM

Is there a way to disable the b2bua feature in asterisk. I would like asterisk to work as a sip server and not be involved in the RTP path between phones. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080104/3716b8e2/attachment-0001.htm

I've been running the 8/1/2004 Head release up until a little over a week ago. I was forced to due to a card failure to upgrade to 1.2.16 without any advance preparation or testing (most of my connections are via satellite to all corners of the globe with high latency). Up until the upgrade I was running with very few issues. Since the upgrade I have been experiencing strange issues

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the