similar to: No matching peer or user based on IP address

Hi list, I try to connect to a GW which have one domain eg sip.mydomain.com and have few IPs related to this domain. I register * to this domain with host=sip.mydomain.com and type=user. So DNS will decide on which IP of my domain I will register (or redirection on the GW side). If an incoming call arrive, I would guess that, as type=user, it will not try to match the IP from INVITE as I

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: >

sip.conf: [imsdroid] type=friend ;;auth=md5 ;;defaultuser=imsdroid secret=mysecret host=dynamic context=cloud-out qualify=60 dtmfmode=auto insecure=port,invite callerid="IMSDroid client" <imsdroid> disallow=all allow=ulaw I've tried with and without defaultuser and secret. sip show peer imsdroid: * Name : imsdroid Secret : <Set> MD5Secret :

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 =>

Sorry for the duplicate post but I have hit a brick wall trying to get this to work. Is there anyone who can help me? I am having trouble trying to register with a voip provider using sip. I am able to connect using xlite softphone. in xlite i use domain/realm: providerdomain.com sip proxy: host.providerdomain.com:9000 this difference in domain and sip proxy host is whats causing

Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd asterisk server. scenario is like this. on my system i am using this peer in sip.conf. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=yes

Asterisk 1.4.32 (Also 1.4.26, 1.4.33) Broadvox ITSP (xxx.xxx.xxx.xxx) Linksys 2102 (yyy.yyy.yyy.yyy) Both peers : canreinvite=yes t38pt_udptl = yes I'm having some trouble getting a T.38 fax call established with Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38 switchover) to Broadvox with the Asterisk server's IP address in the Connection Information (c) instead of

Hi all, I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result. Here is the problem: I have a peer -which is peer AND user- setted up like this [MyPeer] ; type=peer host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 context=from-MyPeer dtfmode=auto disallow=all allow=ulaw,alaw

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

Dear helpeRs, I would like to specify a newline command in R and pass it to latex via Sweave such that it corresponds to latex' \\ command. But that doesn't seem to be possible. If I Sweave the \n character, it just makes a new line in latex but not the \\ command. Is there a way such that the following code would result in latex in blablabla \\ blablabla on different line

-----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

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

hi all, actually i have partially connected the 2 servers but there is a problem. 2 servers A and B server A forwards call to server B without any problem but when i try to forward call from server B to A, server shows the following error on the cli WARNING[7751]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) == Everyone is

On Wed, Apr 15, 2020 at 10:53:05 +0800, 18781374080 wrote: > > > > Hey, guys > > I've been working on whether libvirt supports encrypted snapshots,Here are my versions of libvirt and qemu > > [root@xx ~]# libvirtd -V > > libvirtd (libvirt) 4.5.0 This is too-old encrypted backing files work starting from libvirt-5.10 (but I strongly suggest using at least

Full_Name: Alex Galakhov Version: 2.1.0 OS: Linux (Debian) Submission from: (NULL) (195.19.131.68) After software upgrade source() does not work properly anymore. It completely ignores the chdir= parameter. This is because is.character(file) is always false after the assignment file<-file(file). The fix is (written by hand in diff-like syntax, hopefully, you'll understand it):

Hey, guys I've been working on whether libvirt supports encrypted snapshots,Here are my versions of libvirt and qemu [root@xx ~]# libvirtd -V libvirtd (libvirt) 4.5.0 [root@xx ~]# qemu-img -V qemu-img version 2.12.0 (qemu-kvm-ev-2.12.0-33.1.el7_7.4) Copyright (c) 2003-2017 Fabrice Bellard and the QEMU Project developers 1. assign $MYSECRET to libvirt secret using the secret-define and

Hi, I have a strange problem. At a site where there are 20+ phones, there is one phone that cannot make outbound (to PSTN) calls. Each call is dropped after 20s with "no response to our critical packet". Calls to voicemail and internal extensions work fine. I understand that everything points to a NAT problem, but I don't understand how it could be because: 1) It does not affect

Hi folks, I have been able to migrate a WinNT4 domain to a Samba3 PDC domain using openldap as a backend and smbldap-tools to vampire the WinNT4 domain (pretty much following Samba3 by Example and documentation in smbldap project by IDEALX). Nevertheless, all 600 users migraged from the WinNT4 domain have attributes like these on the ldap database: userPassword: {crypt}x

I have three Sipura 3000 FXO untis for incoming PSTN lines on a small pbx. There is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband

Hello I am running a net work with a NT4PDC a NT4 BDC and attent to add samba BDC. I am following the brief section 28 of the Samba-Howto collection for samba head/3.0. * Adding the samba server as a BDC from NT server manager seems OK * Getting, recording domain sid is ok (see below) > rpcclient NT4PDC -U Administrator password: > lsaquery domain IMAGERIE has sid blablabla > net