search for: sip_peer

I have a dial cmd buried amongst a series of others in a macro like so: exten => s,n,Dial(SIP/1${ARG1}@sip_peer,60,T) Reason for adding a "1" is all the others in the macro don't want the "1" so this was easiest at the time. Now I need to send NA long distance through this macro. All the other dial cmds will just work, but this one is going to try to dial 11NXXNXXXXXX instead of 1NXX...

...=0x8172f28) at sched.c:373 current = (struct sched *) 0x8174868 tv = {tv_sec = 1135275568, tv_usec = 989877} x = 0 res = 1083432672 #5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253 res = 0 sip = (struct sip_pvt *) 0x0 peer = (struct sip_peer *) 0x0 t = 1135275568 fastrestart = 0 lastpeernum = -1 curpeernum = 6 reloading = 0 #6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0 No symbol table info available. #7 0x401ac18a in clone () from /lib/tls/libc.so.6 No symbol table info availab...

Hello I just settup a realtime mysql table for sip_peers. All peers (friends) is autenticateing but when i want to initiate a call between them i got the following error. Someone have some ideea? Thank you. ---<Cut Here>--- pbx*CLI>console dial 1014 == Console is full duplex -- Executing [1014@default:1] Dial("OSS/dsp", &quot...

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

Oh heck. It really looks like realtime has been seriously screwed up. When a call comes in to Asterisk, I can see asterisk executing these queries. SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205' SELECT * FROM ast_sip_peers WHERE name = '2944093' SELECT * FROM ast_sip_peers WHERE name = '2944093' So, the first thing it does is check and see if there are any records in sip_peers where the IP address of the message matches. What happens...

Hello sip show peers does not mark hosts as NAT even though sip.conf and sip_peers table has nat=yes. spitfire*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 5060 Unmonitored 5560/5560 192.168.4.5 D N A 255.255.25...

Out internet connection was out this morning. It seems that the SIP extensions on our LAN were affected. Behavior like: Call comes in over POTS to a TDM400P, there is a delay then before the Cisco 79[46]0's start to ring. If we were lucky enough to get a call through, then we could not transfer the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay

...8 > > tv = {tv_sec = 1135275568, tv_usec = 989877} > > x = 0 > > res = 1083432672 > > #5 0x40927e28 in do_monitor (data=0x0) at chan_sip.c:11253 > > res = 0 > > sip = (struct sip_pvt *) 0x0 > > peer = (struct sip_peer *) 0x0 > > t = 1135275568 > > fastrestart = 0 > > lastpeernum = -1 > > curpeernum = 6 > > reloading = 0 > > #6 0x40024b63 in start_thread () from /lib/tls/libpthread.so.0 > > No symbol table info available. > >...

Hi, Anyone can tell me what this means? [Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter: Inringing for peer 'test-peer' < 0? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090203/a39503b7/attachment.htm

...localhost dbport=5432 dbname=mydb dbuser=pgdbuser requirements=warn [pgwritedb] ;; Connect to mydb2 on another host dbhost=<IP of other db server> dbport=5432 dbname=mydb2 dbuser=pgdbuser2 dbpass=xxxxx requirements=warn In extconfig.conf I configured sippeers => pgsql,general,sip_peers sipregs => pgsql,pgwritedb,sip_regs But asterisk is not using 'pgwritedb' to connect for sipregs, its trying to update 'sip_regs' on the local database. Thanks

Are these both caller id presentation related? If not, which on is currently being used. Finally, is there a "latest" sip_peers table structure to use with 1.8, without the obvious hacks, deprecations. and redundancies? Thanks in Advance, Nick.

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough

Hello everybody! [Introduction] This is a quite long message, but I think the problem is interesting. [The problem] Does anyone know how can I tell Asterisk that a certain user has a certain telephone number (or address)? For example, I have some registered users, but nor the client (X-lite) nor the server (Asterisk) specifies what telephone number has the user. I don't want to

Could someone provide some steps for troubleshooting Realtime? I can't see any signs that it's working. I followed and double-checked a few different guides around the net, but haven't been able to figure it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...ate_peer (p=0x802859a00, expiry=3600) at chan_sip.c:3754 rtcachefriends = 0 #9 0x000000080903dbf2 in register_verify (p=0x802817028, sin=0x7fffff892a30, req=0x7fffff892a40, uri=0x7fffff892e79 "sip:10.3.8.1") at chan_sip.c:11173 res = AUTH_SUCCESSFUL peer = (struct sip_peer *) 0x802859a00 tmp = "<sip:dummyuzer\00010.3.8.1\000\000?\213\002\b\000\000\000\020&\211??\177\000\000\000\000\000\000\000\000\000\000 ", '\0' <repeats 15 times>, " ", '\0' <repeats 15 times>, "\227\027\rK\000\000\000\000\034?\000\...

...2.168.0.2>' failed for '192.168.0.25' - Wrong password [Jul 21 15:40:47] DEBUG[2105]: chan_sip.c:15372 sipsock_read: SIP message could not be handled, bad request: 1832465624 at 192.168.0.25 This error is different to the error that is received if a username that is not in the MySQL sip_peers / sip_users table is specified. Therefore at least the MySQL connection appears to be working. extconfig.conf: sipusers => mysql,asterisk_config sippeers => mysql,asterisk_config I have also tried explicitly adding ',sip_users' and ',sip_peers' to these lines, but asterisk...

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson