similar to: SIP registrations CVS Head

I have noticed the same behavior. I have several different type of phones, and all my Cisco 7960's do not register anymore with the latest CVS release. I have not made any config changes, only the binaries have changed. -brian On Tue, 04 Jan 2005 at 16:38 Andrei (MPI) (asterisk@markovprocesses.com) wrote: > Hello All, > > Sorry if it is known problem. > > I have tried to

Hi, I have several SIP registrations on my Asterisk box. Sometimes, I try to call in the inbound number from 1 and find it doesn't work. When I do sip show registry, it's showing Unregistered (and sometimes there are several which are showing Unregistered). If I type reload, it registers and works fine straight away. Is there something I can set to keep registrations connected all the

The scenario: This is all done SIP with a VOIP provider (have to register to single IP) We have two inbound DID numbers / Accounts. We have to register each individually with the VOIP provider. I'd like inbound from each registered account (DID) to be able to come into a unique PEER or dialplan context. What matters is that the inbound call lands in the context of my choice. I've been

Hi All, Call comes into Asterisk Asterisk answers and Dials SIP Phone SIP phone has call forward enabled to a long distance number Asterisk receives a SIP response 302 "Moved Temporarily" back from phone Asterisk then forwards inbound call to 'Local/number@context' thanks to phone 2 problems with the CDR: 1. intermittent 'bill sec' accuracy, sometimes 0 even when the

In the development tree of Asterisk we've changed two things lately that may affect you: * Asterisk can fail an outbound registration If you enter a register= statement with an incorrect password, wrong hostname or anything else that is wrong, Asterisk will give up registration after 10 attempts. If you do "sip show registry" you will see the registration status as

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Has anybody successfully configured asterisk to use Galaxy Voice for inbound / outbound calls and what is required to make this work? Dave --------------------------------------- Dave Tipton dave@lodave.org 817-858-9841 Home 469-223-8506 Cell

I finally decided to get a little source code dirt under my fingernails tonight and dig through chan_sip.c to understand how registrations are currently implemented. The hope is to perhaps at least seed some ideas about how to make registrations to a server name, which resolves to multiple IPs, either attempt each IP in the order they're returned by dns, or, simply attempt to register with

Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and

Hello list I've been implementing an asterisk based cluster, and are having grave problems with SIP. My current implementation monitors registrations through the Asterisk Manager interface, but it seems to not register all registrations. Because of this, ive been looking at DUNDi, to implement the cluster. Has anybody done this with success? It seems that it uses the extensions instead of

Hi, For various reasons a customer of mine is moving from a SER-based to an Asterisk-based installation, mostly because of problems with SIP devices behind NAT trying to reach each other and because it's easier to do accounting when all calls go through Asterisk (canreinvite=no is the idea). The database-based SIP registration mechanism of Asterisk seems to have one shortcoming - it

Hi, I had a Trixbox 1.4.18 that I "yum update"d to 1.4.22.1. Now, I seem to have a huge problem with phones not staying registered (registrations worked perfectly at 1.4.18). I phone will register the first time I plug it in, and then once you make a call and hangup (or sometimes even during the call) all the lights will go orange meaning a lost registration. Every so often the lights

Hi All, I am new to this community and just subscribed. We have Asterisk running in production but I could not find out in documentation as well as web that how many maximum number of registrations an Asterisk Server can support. We have it on a 1.4 GHz Processor, 2 GB RAM and 40 GB HDD IBM Server. Please suggest urgently. Thanks. Best Regards, ------------------------------------- Abid

Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part

Hello everyone I want to know if it is somehow possible for asterisk to consider new registration attempts instead of matching them with old nonce Correct auth, but based on stale nonce received from '"test" < sip:3247 at 1.1.1.1>;tag=79a401979bffd0d9o0' I see messages like the one above, I understand it is because of existing expire value but would like the previous

Hello list, I'm using asterisk 1.6.2.11 with realtime SIP (mysql DB). I notice that when the SIP peer registers, the fields 'fullcontact', 'ipaddr', 'port', 'regserver', 'regseconds', 'lastms' are filled with values. But after a while, these fields become empty. Asterisk CLI shows : asterisk*CLI> sip show peers Name/username

After fighting with a "Unable to create/find channel" [1] [2], I gave up on my previous installation and rebuild my asterisk from CVS-Head. I guess the Debian package available today is broken somewhere (after a previous broken release made with an old libpri package), but now I'm having another issue with my 7960 registration (SIP v. 7.1). The call is being (silent) rejected by

Hi all, Is there anyway i can prevent multiple sip registrations from different IPs using single username in asterisk. Does asterisk provide any aid in this respect? As far as my knowledge is concerned i dont think there is any support for this in asterisk, so i think i'll have to makeup a script which sniffs sip packets coming for asterisk and detect for multiple register requests coming from

I tried using CVS HEAD 08-05-2004 for zaptel/asterisk last night but I'm having problems with zaptel. The last CVS HEAD I was using was CVS HEAD 07-31-2004. (BTW... I'm a CVS moron... what command would I use to get back to that version of HEAD because like an idiot I forgot to make a backup copy of the CVS HEAD on my system before I install from the new CVS HEAD. I tried cvs co -D

I have looked into this callerid problem now for a few hours. 1) Caller id on a sipura-2000 now shows: cidname 2000 Where cidname is the new outputted formate from the cid_rewrite agi script and 2000 is the exten number. In looking at the Dial() application, option "o" 'o' -- Original (inbound) Caller*ID should be placed on the outbound leg of the call