similar to: Several doubts on Asterisk as an UAC

Displaying 20 results from an estimated 10000 matches similar to: "Several doubts on Asterisk as an UAC"

2007 Aug 07
1
Use of context=... in [default] section of sip.conf
Hi, If I have [myprovider] section with context=something. When I do an outgoing call by using Dial(SIP/myprovider/464646)", does context=... affect anything? As I understand it, it only affects incoming calls, but I might be wrong. Another thing, the setting of context=... on [default] section will affect all [provider] sections without context=..., right? What if I don't specify any
2010 Dec 14
1
Asterisk + VOSP account working configuration?
Hello I'm having a difficult time finding precisely what to put in sip.conf and extensions.conf (and possibly other files) to get a working configuration to connect an Asterisk (1.4) server to a VoIP provider with the Asterisk server + SIP clients located in a private LAN behind a NAT router: http://img560.imageshack.us/img560/3749/asterisknat.png Would someone have a full, direct (ie.
2007 Nov 20
0
MediaHandling--Help Required
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
2005 Mar 19
1
Asterisk Quicknet FWD Problem - no path to translate from Phone/phone0 to SIP
I cant seem to be able to figure this out. As much as I can tell it is a codec problem. I can dial out to 612@fwd.pulver.com and the "Call Me" test there rings my phone. However when the callee endpoint answers, there is a failure to translate: Outgoing Call for 612 612 is not a local user -- Called 612@fwdpulvercom No path to translate from SIP/fwdpulvercom-dd5a(2) to
2005 Feb 14
0
Asterisk as SIP UAC !!!
Hi gentleman I've configured SER to forward every call starting with sip uri request "1" to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it call to my other SIP Provider outside my network, sending username and password for authentication. I've read at www.voip-info.org some articles but found none that could suit to my needs, but yet I've found an
2020 Jan 16
1
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to
2007 Dec 28
1
sip.conf & realtime
Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]
2020 Jan 20
0
Asterisk16 - PJSIP - Error 401 on outbound registration
On Sun, Jan 19, 2020 at 10:45 AM Administrator <admin at tootai.net> wrote: <snip> > It become stranger and stranger: on one of the register peer we receive in > asterisk: > > *CLI> [2020-01-19 15:23:18] WARNING[17469]: > res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal > response '401' received from
2007 Dec 29
1
Realtime & sip.conf
Hi - I'm looking into realtime and I'm having a bit of a problem with the SIP part. My review of the posts seems to indicate that I should use realtime static for the [general] part of my sip.conf including the registration commands: register=><did>:<secret>@<domain>/<did context> and use realtime realtime (funny name!) for peers and friends: [myprovider]
2005 Jul 18
0
IAX register confusion
I have been unable to understand the connection between an IAX registration for dynamic IP assignment and and the host definition. I have signed up with an ITSP for a DID. My ip is dynamic and although I have a dynamic DNS name, we are registering and outbound works fine. I'm at a loss to understand the relationship between the registration and the [section] definition in iax.conf that will
2020 Jan 19
2
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 19/01/2020 à 00:31, Joshua C. Colp a écrit : > On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > > Le 17/01/2020 à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange
2006 Nov 15
1
Attempting native bridge of
I have the following scenario: g729 gsm UAS <-----------> * <-----------> UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ?Am I wrong? The UAC and UAS are
2012 Jan 14
1
Asterisk as UAC: How to put call OnHold
Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the
2017 Apr 03
3
Define SIP fromuser field in Dial()-command
Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at
2012 Dec 11
0
dovcot+offlineimap+gmail: INBOX empty, mail doubled
Hello I have used for years offlineimap-6.2.0 dovecot-imapd 1:1.2.9 in Kubuntu 10.04 Together with a imap server of my university. Yesterday we switched to google, which I know, is not a real imap server. I have 2 problems and the first I think has to do with either an old version of offlineimap or a bad configuration file: - All folders are downloaded+ Gmail[All Mail] so doubling
2007 Jul 12
0
No subject
help me in another issue related also to registering asterisk with another softswitch: A) If nat=yes, then I have to set canreinvite=no to be able to register, correct? B) In case of using firefly softphone, how it possible to set it to have nat=yes (at the firefly it self and not at the sip user configuration section)? As most of the sip endpoint give an option to set nat=yes and so on, how it
2010 Sep 15
2
Digest Username/auth name mismatch‏
Hi I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [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 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for
2006 Feb 12
0
[ANNOUNCE] PKCS#11 support in OpenSSH 4.3p2 (version 0.07)
Hello, The version 0.07 of "PKCS#11 support in OpenSSH" is published. Changes: 1. Updated against OpenSSH 4.3p1. 2. Ignore '\r' at password prompt, cygwin/win32 password prompt support. 3. Workaround for iKey PKCS#11 provider bug. 4. Some minor cleanups. 5. Allow clean merge of Roumen Petrov's X.509 patch (version 5.3) after this one. [[[ The patch-set is too large for
2004 May 10
0
polycom ip 500 registration problems
hello all, I'm having problems getting my polycom soundpoint ip 500 working, and was wondering if anyone would be willing to share their config files with me (the polycom configs). I have managed to get my boot server up and running, and the phone successfully updated its ROM, and downloaded the config files i have put together for it (the display shows correctly but the line won't
2004 Nov 26
1
direct asterisk to asterisk SIP calls without external SIP provider
Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line