similar to: Asterisk as SIP proxy?

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will not be charged for Meaning less network problems Because there is no way asterisk knows about

Hi, I just ran into what seems to be an issue on re-invites. I'm not sure if it's a bug or as designed, so I thought I'd ask the question. Here's my setup: - Asterisk 1.8.13.0 - Phone A: Polycom ip331, only allowed to use ulaw, canreinvite=yes - Phone B: Polycom ip330, only allowed to use alaw, canreinvite=yes Phone A calls the extension of phone B. After the normal call setup

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number

Hi all, Hope somebody can help-I really am stumped as to why this won't work. I realise that this isnt an Asterisk problem (Please dont bash me on the list) and I have emailed the SER list but I havent received a reply and maybe someone on this list can help...Once this problem is solved I am going to use Asterisk for voicemail etc with SER (I have it set up) I currently have SER set up and

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

I have been given the following settings for connecting to a voip provider. The names of the fields match my snom phone, and when configured, the phone both makes and recives phonecalls without issue. I am trying to put the same values in asterisk, but there seems to be one field that doesn't seem to exist in asterisk - that of outbound proxy all suggestions welcome SIP headings account

I must be doing something incorrectly, or something is wrong with ATA-186 reINVITEs in SIP. Perhaps someone more enlightened than me can lend me a hand. I have been attempting to get two SIP phones to reINVITE to each other, and I've been unable to think of or uncover the correct method. The calls always go through the Asterisk server, no matter what I try. I've simplified things

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a proxy, but when connecting SIP to SIP it works only as 'SIP registrar' forwarding SIP requests to client. Is it possible to make Asterisk work as a 'proxy' so that any incoming calls would be ade to Asterisk and then internally forwarded to receiver?

> Hi All, > > We are using asterisk for testing our home gateway setup. > We have implemented Call Hold feature in our application. > In our Application we have written code in such a way that for an INVITE > for > putting a SIP phone on HOLD will contain connection address "0.0.0.0" in > the SDP message. > We expect the same connection address i.e

I'm using the latest SIP firmware from Avaya. The phone receives the 46xxsettings.txt OK, and then after entering extension and password it goes to the home screen saying 'Registering'. When I check options->ViewIPSettings->IPAddresses on the phone, the registrar and SIP Proxy fields are blank. I have both lines: SET SIPREGISTRAR "67.1XX.XX.XX" and SET

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Hopefully this is an easy question.... I''m using a leaf router (bearing) running shorewall. Three interfaces net, loc, and dmz. Only one computer in the dmz and its being proxy arp''d. External and internal (net and loc) can reach the dmz but the dmz cannot reach the isp''s gateway and beyond, but can reach a system adjacent to the firewall.

Hi, I'm using opensips as the registrar server for my users. I am redirecting calls going out to pstn to my asterisk server. call flow is basically: ua --> opensips server --> * server --> sip gateway provider if (uri=~"sip:00[0-9]*@sip\.myserver\.com") { xlog("L_INFO", "Call to PSTN\n"); #strip(2); #prefix("011");

Hi all, So the scenario is: A -> Asterisk -> B after B send back 200 OK Asterisk is answering the call to A. Directly after the Answer Asterisk generates a ReInvite to A and the only difference between the 200 OK sdp and the reInvite sdp are the offered codecs which are forwarded from B to A. Here i do not understand why this could not be done in the 200OK to A? As far as i understood

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

Hi All, I'm having a really weird can reinvite issue. I've been banging my head around on this for days now.. I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 172.20.0.11 is a hosted box and serves multiple offices 172.20.2.5 is a box on site at a customer's office. A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a phone at 172.20.2.80 via server