similar to: SIP registration

Hello All, For some odd reasons my Asterisk is keep on loosing registration of my SIP devices. On the SIP device it shows I am RESISTED but when I do "sip show peers" it shows my sip endpoints are "UNREACHABLE". And it keeps on flapping "Peer '9099993456' is now UNREACHABLE!" and "Peer '9099993456' is now REACHABLE!"... I changed my

Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using

I'm having trouble with Polycom 501 phones that asterisk forgets how to reach them. /etc/asterisk/sip.conf: [general] context=default MusicOnHold=default port=5060 bindaddr=0.0.0.0 srvlookup=no;yes language=en dtmfmode=rfc2833 maxexpiry=600 defaultexpiry=120 [502] type=friend username=502 secret=pass host=dynamic mailbox=502@rm callerid= "Operator" <502> context=rm

Hello is it possible to determine the SIP.conf parameters 'defaultexpirty' and 'maxexpiry' on a peer basis ? My default value is 300 seconds, but some specific SIP-clients can only send a SIP REGISTER every 3600 seconds. In current configuration these SIP peers now become "Unreachable" after 300 seconds. Or is there another way to differentiate ? Kind regards.

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hi, I've got several trunks in my 1.6.0.1 setup. One of them is asking for 1800 sec registrations. You can provide this value setting defaultexpiry to 1800 in sip.conf but how can you specify this duration to this specific trunk and not affect the others ? An option to register statement in sip.conf would be perfect ... Regards -------------- next part -------------- An HTML attachment was

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

Hello list, I have in sip.conf : /maxexpiry=60 ; Maximum allowed time of incoming registrations ; and subscriptions (seconds) minexpiry=60 ; Minimum length of registrations/subscriptions (default 60) defaultexpiry=120 ; Default length of incoming/outgoing registration ;-----------------------------------------

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hello all, I'm having a quite simple configuration like: SIP provider <=> asterisk SIP <=> lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net

Hi, This is the first time I experience this problem with Asterisk: all of a sudden SIP registrations stopped working. Active calls kept working but new calls could not be established (I did NOT perform a "graceful restart"). Besides, would a "restart gracefully" actually deny SIP registration? I did not have a network issue because killing asterisk and starting it again

We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to 999@pbx.itsp.com REGISTER attempt 2 to 999@pbx.itsp.com Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why

Hi, I am registering my Asterisk boxes to a SIP provider for outgoing calls. My "outgoing" dialplan context tries to dial out in sequence, starting with the SIP provider then ISDN lines and finally analog lines. So the idea is that if the SIP trunk fails then all calls are dialed out via ISDN and analog. I noticed however that if I switch my DSL connection off (ie. no internet access

Hi, does any know why? i can make call out with my asterisk and voipstunt but i can't get call in on my voip in number i get rejected. if i use Sipura without asterisk i get in calls here is my sip.conf ---------------------------------------------- [general] useragent=nedi port=5060 context=default ;tos=lowdelay disallow=all allow=ulaw allow=alaw allow=gsm allow=g726 language=de

I am having trouble setting this system up and wonder if some one help me. Does anyone know what is missing if anything to get 2 phones on my asterisk home server to be able to call each other. I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2 extensions 5060/5061, this is on the lan side of my gateway/router WRT54G 192.168.1.1 BusyBox v1.00 (2006.11.07-01:40+0000)

I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for my problem Hello, During a call with canreinvite = no, at the beginning of the call I lose 2 seconds of audio. is obvious when I call autoattendant. schema: SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1) --> Operator SIP capture of voip1: - Executing [0825387205 at

Apologies if this is a repeat: I trawled through the archives and couldn't find a reasonable answer, so I'm asking here. I have an Asterisk install connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case, Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT device that connects to the Asterisk install, and using this setup I've been pretty

I'm trying to get a Zultys Zip 2 phone working with Asterisk. The phone seems to be failing registration (see sip debug output below). However, I can place calls TO the Zip2 from other SIP phones (Grandstream BT-101, Xten X-Lite, and eStara Softphone) and from Nortel PBX extensions coming in to Asterisk over a PRI T1. The problem is that I cannot dial any extensions from the Zip 2. Any

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hello All, I am trying to get some ATA 188 units to register with my Asterisk box over SIP. I continue to get the same "401 Unauthorized" Error when they try to register. If I turn Sip registration off, I can use the phones without any problems with a static IP assigned in my sip.conf file, but I can't get the second phone port working. I've set up two separate logins both