search for: srvlookup

I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues (Asterisk stops responding). I use VoIP Buster and in sip.conf I use sip1.voipbuster.com. When I do sip show peers in CLI I get voipbuster/tomo 194.221.62.207 5060 OK (27 ms) And when I ping sip1.voipbuster.com [root@tom...

Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify: * srvlookup=yes|no * pedantic * canreinvite=update|yes --"update" seems new Being curious, especially for "srvlookup" functionality... /O

...s18abefe8 Can someone help me out of this? I have Asterisk 1.2 on the Ubuntu 7.04. Outcoming and internal calls functions well. Thanks sip.conf: [general] callevents=yes register => username:password at sip.mujtelefon.cz/username [100] callerid=100 secret=password type=friend context=internal srvlookup=yes type=friend qualify=yes nat=no host=dynamic canreinvite=no context=internal call-limit=1 [101] callerid=101 secret=password type=friend context=internal srvlookup=yes type=friend qualify=yes nat=no host=dynamic canreinvite=no context=internal call-limit=1 [TRUNK-587207103] type=friend context...

Ok we have found a better solution. Put everthing back the way it was and make sure that you have this line in your general section of you sip.conf file: srvlookup=yes We have added a SRV entry in the correct place now. So everyrthing should go the correct servers. -james --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.726 / Virus Database: 481 - Release Date: 7/22/2004

...ea codes that are local to me in Dallas. Can anyone help me? here are my config files... - sip.conf - [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP bindaddr=10.1.1.200 ; IP address to bind to (0.0.0.0 srvlookup=yes ; Enable DNS SRV lookups on register => xxxxxxxxxx:secret@sip.broadvoice.com externip = 209.30.232.185 ; Address that we're going to put in outbound SIP messages localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 [broadvoice] type=peer nat=yes username=xxxxxxxx (phone number) se...

...ncomingcalleridonzaptransfer=yes echocancel=yes echocancelwhenbridged=yes faxdetect=yes rxgain=1.0 txgain=21.0 callgroup=1 group=1 usecallerid=yes callerid=asreceived cidstart=ring hidecallerid=no immediate=no pickupgroup=1 ;context=incoming channel => 1-4 Sip.conf [general] srvlookup=yes ;allows DNS lookups of server names naxexpirey=180 defaultexpirey=160 context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) tos_sip=cs3 tos_audio=ef ; bi...

...the call, or place the call on hold, or park the call. Outbound calls seemed to have a delay between the time they were dialed at the SIP phone and when they were connected. I know this has been brought up before, in fact there is a bit of a discussion going on now about DNS SRV (in sip.conf, set srvlookup=no, or put all the phone ip's on /etc/hosts). But what is really causing the issue here? Yes, it is DNS, or something related to DNS, but why does that have anything to do with * trying to make a phone ring on the LAN? I would think that by using qualify=yes for any outbound voip trunks we avo...

...e thing more if i try to use playback application for playing some sound file it is also working (like exten => 500,1,Playback(demo-abouttotry) this is working). here is sip.conf //////sip.conf////////////// //////////////// [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes allow=all nat=no [6000] type=peer host=dynamic context=default canreinvite=yes allow=all [1000] type=peer host=dynamic secret=1000 canreinvite=yes allow=all __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around...

...are on a private net (10.9.8.xxx), but as it's the Asterisk box that's doing the NAT'ing, I used nat=no; I presume that's correct. eth0 has address 10.9.8.1, while eth1 has a global internet IP address. Cheers, Nikhil. ----- Extract from sip.conf: [general] context=incoming srvlookup=yes realm=nikhil-nair.net ; Various "register=>" statements, not relevant to the local phones [101] ; Aastra 9112i at 10.9.8.101 type=friend secret=... qualify=yes ; Qualify peer is no more than 2000 ms away nat=no ; This phone is not natted host=dynamic ; This device registers wit...

.... > By this I mean it does not answer call coming in from the gateway > (which is on the local LAN) and you can't even reload it - just hangs > there. If I change the DNS setting in resolv.conf to something else > which is reachable all is well again. > > I have tried setting srvlookup=no in sip.conf but it made no difference. > > Does anyone know how I to make Asterisk continue working for local LAN > users/gateways when a DNS server is not reachable? Try to use bind on the * Machine and configure it as a caching only nameserver. Hope, this helps Regards, Guido Hec...

...or 'XXXXX@sipgate.de' timed out, trying again (Attempt #n) I looked at the sip debug stuff, and all I can see is my asterisk sending the registration packets, but no answer is received. Here's the relevant parts of my sip.conf: [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes tos=0x18 checkmwi=10 videosupport=yes allow=all relaxdtmf=yes rtptimeout=60 rtpholdtimeout=300 register => XXXXX:pass@sipgate.de/XXXXX ;XXXXX == sipgateid [XXXXX] type=friend insecure=very nat=yes username=XXXXX fromuser=XXXXX fromdomain=sipgate.de secret=pass host=sipgate.de qualify=yes...

...ddress to bind to externip = 69.132.68.17 ; Address that we're going to put in SIP messages if we're behind a NAT localnet = 192.168.1.0 ; Internal NETWORK address localmask = 255.255.255.0 ; Internal netmask context = default ; Default for incoming calls ;srvlookup = yes ; Enable SRV lookups on outbound calls ;pedantic = yes ; Enable slow, pedantic checking for Pingtel ;tos=lowdelay ;tos=184 ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming...

...experiencing a drop in call. It does not have a certain time, it is random. The audio is flowing normally and the call is dropped. Has anyone ever experienced this? My settings changed below: allowoverlap = no udpbindaddr = 0.0.0.0 tcpenable = no tcpbindaddr = 0.0.0.0 transport = udp, ws, wss srvlookup = yes directmedia = no rtcachefriends = yes externaddr = my ip address externhost = my domain address ;   foo.dyndns.net; refreshed periodically externrefresh = 180       localnet = 172.31.40.21 / 255.255.240.0; AWS NETWORK       localnet = 192.168.0.0 / 255.255.0.0; RFC 1918 addresses     ...

...ay's case to Cuba). Here's a copy (slightly modified) of my sip.conf: [general] context=default ; Default context for incoming calls videosupport=yes rtcachefriends=yes autocreatepeer=no t38pt_udptl=yes allowoverlap=no udpbindaddr=0.0.0.0 srvlookup=yes ;pedantic=yes disallow=all allow=alaw allow=ulaw allow=speex [1001] type=friend username=1001 secret=blah subscribecontext=default regexten=1001 callerid="blah" <XXXXXXXXXX> host=dynamic nat=yes canreinvite=no mailbox=1001 at default registertrying=yes [testuser] type=friend...

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

...361355,3,Goto(testdtmf|s|1 ;Ring on Elle mobile phone. ;exten => s,1,Answer() ;exten => s,n,Dial(SIP/17037171234,150,r,t,) [general] register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146 registertimeout=20 context=incoming allowoverlap=no bindport=5060 bindaddr=192.168.1.10 srvlookup=no ;context=incoming ; The SIP provider [voipvoip.com] canreinvite=no username=1112530146 fromuser=1112530146 secret=albany!@#123 context=incoming type=friend fromdomain=sip3 at voipvoip.com host=69.90.209.57 dtmfmode=rfc2833 disallow=all allow=alaw allow=ulaw nat=force_rport insecure=port,invite...

...the top part of my sip.conf ;_____________________________________________________________ ;sip.conf ;_____________________________________________________________ [general] port=5060 bindaddr=0.0.0.0 externip=XXX.XXX.XXX.XXX localnet=XXX.XXX.XXX.XXX/255.255.255.248 canreinvite=no tos=reliability srvlookup=yes disallow=all allow=ulaw dtmfmode=rfc2833 nat=yes ignoreregexpire=yes I know it has something to do with the NAT because if I plug my Polycom directly into my cable modem, thus making it sit on the Internet and have a real IP, everything works just fine. I am curious what I am missing. Thanks...

...solve this problem for more than one month. I have included the sip.conf and the extensions.conf below. [SIP.conf] ; SIP Configuration example for Asterisk [general] context=incoming allowoverlap=no bindport=5060 bindaddr=0.0.0.0 localnet=192.168.5.0/255.255.255.0 externip=a.b.ccc.dd srvlookup=yes allow=ulaw allow=alaw [incoming] type=peer nat=no canreinvite=no host=xx.y.z.aaa qualify=yes dtmfmode=rfc2833 context=default [extensions.conf] [general] static=yes writeprotect=yes clearglobalvars=no [default] include => customer exten => h,1,Hangup exten =&gt...

...therlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my sip.conf looks like this: [general] context=incoming-j canreinvite=no dtmfmode=inband qualify=yes srvlookup=no disallow=all allow=alaw allow=ulaw allow=g722 allow=g726 allow=g729 insecure=port,invite register => <telno>:<password>@sip.xs4all.nl/<telno> Does anyone know of any new variables that have been introduced since Asterisk 1.6.2.9, that apply here and might be causing this p...

...the problem? The Inbound call works fine. Here is my sip.conf [general] context = demo ?; ? ? ? ? ? ? ?Default context for incoming calls bindport = 5060 ?; ? ? ? ? ? ? ?UDP Port to bind to (SIP standard port is 5060) bindaddr = 0.0.0.0 ?; ? ? ? ? ? ? ?IP address to bind to (0.0.0.0 binds to all) srvlookup = yes ?; ? ? ? ? ? ? ?Enable DNS SRV lookups on outbound calls context=incoming disallow=all allow=ulaw allow=alaw allow=g729 allow=g723 externip=72.220.28.226 localnet=192.168.0.0 nat=yes maxexpiry=15 minexpiry=14 ;rtautoclear=no ;autofallthrough=yes register =><did>:<password>@69....