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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 offic...

...hat's the same if we call remote (or receive) from an analog or iax phone. Here's a snippet of my sip.conf: ; ; SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 0.0.0.0 context = local tos = lowdelay disallow = all allow = ulaw ;local phone definition [200] accountcode=localphone mailbox=200 type=friend secret=secret username=200 host=dynamic callgroup=1 pickupgroup=1 ; remote phone definition [250] accountcode=remotephone type=friend secret=XXXXX nat=yes username=250 context=local reinvite=no disallow=all allow=g729 canreinvite=no host=dynamic qualify=1000 callgroup=1 pic...

...hat's the same if we call remote (or receive) from an analog or iax phone. Here's a snippet of my sip.conf: ; ; SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 0.0.0.0 context = local tos = lowdelay disallow = all allow = ulaw ;local phone definition [200] accountcode=localphone mailbox=200 type=friend secret=secret username=200 host=dynamic callgroup=1 pickupgroup=1 ; remote phone definition [250] accountcode=remotephone type=friend secret=XXXXX nat=yes username=250 context=local reinvite=no disallow=all allow=g729 canreinvite=no host=dynamic qualify=1000 callgroup=1 pic...

...-- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Zap/g4/w5551234") in new stack -- Called g4/w5551234 -- Zap/4-1 answered Zap/2-1 -- Attempting native bridge of Zap/2-1 and Zap/4-1 -- Hungup 'Zap/4-1' == Spawn extension (localphone, 5551234, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' *********************************************************** So one can presume that asterisk is hanging up the PSTN line, for some reason. By why? There is clearly not silence on the line. Any thoughts? joe a.

...8) currently in production. Both systems are on VPS with public IP addresses. Goals for the new system include: HD (g722) connections on internal calls, Asterisk only proxies audio when necessary, no unwanted transcoding. For initial testing, I've set up two Yealink T26P extensions and one Localphone trunk. Internal and external calls work, except for the problems above. The extensions are behind a NAT, but are set up with STUN, unique SIP and RTP ports, and proper forwarding. The router handles hairpin connections properly. When registered to the old system, calls between the test extensio...

...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-9...

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 mailbox #) The output from sip set debug is attached, as captured earlier by the script command. Asterisk version 1.4.13, Debian GNU/Linux Sid (up to date); this pho...

...antispam: run program failed with exit code -1 This is probably a permissions issue but I'm not able to debug it. Can anyone offer me any clues as to what I might be doing wrong or how I might fix it? Thanks, Anthony -- Anthony Papillion Phone: 1.918.631.7331 VoIP (SIP): 1259010 at localphone.com XMPP Chat: cypher at chat.cpunk.us Fingerprint: 65EF73EC 8B57F6B1 8C475BD4 426088AC FE21B251 PGP Key: http://www.cajuntechie.org/p/my-pgp-key.html