similar to: asterisk <-> SIP provider, two way connection

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Am 13.06.2020 um 08:28 schrieb Luca Bertoncello: > Hi! > > I have a Asterisk installation to manage my phones at home (provider is > Deutsche Telekom). > It works, but very often the voice is "broken"... > Yesterday during a call it was very difficult to understand what my > partner sayd... > > It can NOT be a problem of other downloads/uploads, since in that

If one of the authors is listening: http://www.asteriskdocs.org/html/apas02.html lists usereqphone 2 times. One of the entries should really be useragent. And the example for usereqphone is wrong. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? ->

Hi guys, I create a topology like fellow: /****** /************ /*********** * GK *-----------* Asterisk *---------- Sip Prov * ******/ ************/ ***********/ | | | | | | H.323

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, I'm looking for a network management software. And as the network grows it clearly becomes that manual notes is getting too tedious. Also an integrated troube ticketing systemm would be great. Any reference is really appreciated. Thanks. - -- Fajar Priyanto | Reg'd Linux User #327841 | Linux tutorial http://linux2.arinet.org 13:10:54

http://bugzilla.netfilter.org/show_bug.cgi?id=704 CZ <huangj at qualcomm.com> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|RESOLVED |REOPENED Resolution|FIXED | --- Comment #4 from CZ <huangj at qualcomm.com>

Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? Or any Nortel "sip" products? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back

Hi everyone, I am starting to work with PJSIP on release 12.1.0.rc3. I used to have Asterisk 1.8 with the regular sip channel. I was using the usereqphone settings in order to set user=phone on from and to URIs. Is there a similar config in PJSIP? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear sirs ! I have the only troube with PCAD under wine - PCB router shows all traces 2 times thicker than real. (any version of wine in any configurations) It is possible to download 30 day trial for experiments. There are some number of noncritical bugs, but trace width is most significant. This feature prevent me from using pcad under wine. Will anyone solve this problem ? Will it ever

Hi there, I'm tring a basic asterisk settings. I have a asterisk 1.2.7.1 running on a I have a net with two computers and a router. The router IP in the local net is 192.168.1.1, The first pc has IP: 192.168.1.3 name datile3 . SO GNU Linux. the second pc has IP: 192.168.1.4 name fissun . SO GNU Linux. On datile3, it runs a softphone kphone. From this I want to call the external world. on

i had a sip invitation problem with my voip provider and here the message that was shown : Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:XXXXX@195.112.214.99>;tag=as7b43dfbd' -- SIP/callshop-3fcc is circuit-busy == Everyone is busy/congested at this time -- Got SIP

Hello list, I would like to seek your expert opinion on a setup I am trying as part of my research. I have not been able to successfully make a call so far. In my setup, I use two laptops that are interconnected by means of a stand-alone IS1581 switch. Thus there is no LAN involved. I have assigned static IPs to the two laptops, say 10.0.0.1 and 10.0.0.2. I have installed Asterisk 1.6.2.6 and

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

I recently came into this troube, when i use some acl edit script to update ACL information for the files hosted in samba server, the winbind server suddenly got panic, the related log file looks like this: [2006/04/21 18:24:02, 10] libsmb/smb_signing.c:simple_packet_signature(270) simple_packet_signature: sequence number 5112 [2006/04/21 18:24:02, 10]

i have opened an account with callshopcompany,and when ive tried to send calls by the sip i had a message show an asterisk invitation problem i had these sip configuration: sip.conf [callshop] type=peer host=sip.callshopcompany.com username=XXXXX secret=XXXXXXX Then i tried to add these lines and it worked : sip.conf [callshop]

Has anyone had any luck getting an asterisk box to talk to a Nortel C15K softswitch? I've been playing with it for several days and can't seem to pass calls either direction. I know that whike the Nortel says the C15K speaks SIP, it really speaks nortel's implementation of SIP, but I thought I could get it to at least pass simple calls back and forth to an asterisk box. Right now,