search for: localphones

Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, i know i am asking a lot of questions lately in this forum..sorry about that first of all. :) Ok, here is the deal.. I am trying to make a hybrid system with an ericsson MD110 and asterisk. Internally we have 4 digit phone extensions on ericsson.. and so in asterisk. So, what i want to do is to call pbx side without adding 9 or etc to the begining of the number from asterisk clients.. For

.... ] DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping retransmission on 'b5288d54-a46c-9e16-ff7c-ec43221a71b2@192.168.1.190' of Response 53320: Found [ ... ] DEBUG[5126]: File chan_sip.c, Line 991 (find_user): Call from user '17476691152' is 1 out of 0 Looking for 2 in localphones DEBUG[5126]: File chan_sip.c, Line 3369 (build_route): build_route: Contact hop: <sip:17476691152@192.168.1.190> -- Executing Playback("SIP/17476691152-a52e", "publicar-extbusy|skip") in new stack *CLI> some time ... a few seconds No such command 'some' (...

...to my provider because they don't offer it and my handsets ( snom 300 ) also don't support IAX so it's all SIP. The configuration is a follows Asterisk PBX 10.202.17.217/24 ------>| 10/100-Switch |-----> Firewall1 pfsense X.Y.Z.250 -------->ITSP Sip Porvider public internet LocalPhones 10.202.17.1-25/24 -_---->| 10/100-Switch |-----> Firewall2 Watchguard ----->ISP internet Connection <-----Firewall3 | remote office | ----Remote User Phone 192.168.97.74/24 There is a Lan2Lan VPN tunnel between the Firewall2 and the Remote Office Firewall3 I can Ping the remote office...

Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones. I have 2 1 sip phone that's outside in the "world", and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. The hangup does *not* occur when entering the numbers, as one might initially expect, but after the called end begins to

I am unable to get re-invite to work on a new system. Also, unwanted transcoding is occurring on PSTN calls. The new system (FreePBX 2.11.0.37, Asterisk 11.9.0, CentOS 6.5) will eventually replace an old system (FreePBX 2.8.1, Asterisk 1.8.7.2, CentOS 5.8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on

...ng Playback("SIP/17476691152-7158", "extbusy|skip") in new stack -- Timeout on SIP/17476691152-7158 == CDR updated on SIP/17476691152-7158 -- Executing Hangup("SIP/17476691152-7158", "") in new stack == Spawn extension (localphones, t, 1) exited non-zero on 'SIP/17476691152-7158' DEBUG[15376]: File chan_sip.c, Line 1068 (sip_hangup): find_user(17476691152) - decrement inUse counter DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping retransmission on '75057cca-9a...

Well, I think I discovered even further why there is no ringback tone available. The following message, is displayed on the console in asterisk. ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Looking more into it, I found that it was related to loading tones for a particular zone. The message is printed

[general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to A realm=192.168.0.2 context = default ;Default for incoming calls [5549] disallow=all allow=ulaw allow=alaw allow=gsm type=friend ;(inbound and outbound calls accepted) secret=localphone ; obvious password for testing host=dynamic callerid=Jason White <5549> dtmfmode=auto mailbox=5549 ;(Asterisk VM-system's

Hello Everyone, I'm running the antispam plugin on Dovecot 2.0.19 on Ubuntu Server 14.04 and I can't seem to get it to work. In the IMAP section of dovecot.conf I have the following lines: protocol imap { mail_plugins = $mail_plugins imap_quota imap_acl antispam # mail_plugins = $mail_plugins imap_quota imap_acl imap_client_workarounds = tb-extra-mailbox-sep # Maximum