similar to: The same SIP problems...SORRY!

I write the email again, the third time!!, cause the other two ones, I have had problems while sending them. I hope this time it works. Here is the email again: Hi (and sorry) everybody I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! 1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have

Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My

hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type

Hi, I encountered a problem when I joined a FreeBSD machine to a windows = 2003 AD domain. I passed following steps successfully: 1.#net ads join =A8CUAdministrator =20 2.#wbinfo =A8Cu 3.#wbinfo =A8Cg But when I check: #id Domain\\username (I can find this username using command = =A1=B0wbinfo =A8Cu=A1=B1) Id Domain\\username: no such user =20 I found: After restarting samba, there

Here is the CS1000 log. Again, the CS1000 using SIP accepts incoming calls just fine. However, using outgoing call files the CS1000 is hanging up after I answer the call. I dont know why? Thanks, for any assistance. Jerry my sip.conf entry is: [Nortel] type=friend dtmfmode=rfc2833 username=XXXXXXXXX disallow=all allow=ulaw allow=alaw

From a 'sip debug': Sip read: SIP/2.0 100 Trying From: "Evert"<sip:[username]@[my ext. IP]>;tag=as6e18534e To: <sip:[dialled number]@[SIP server of VoIP provider]> Call-ID: 6cbf41c25281f08b2e7bbc5043061975@[my ext. IP] CSeq: 102 INVITE Via:SIP/2.0/UDP [my ext. IP]:5060;branch=z9hG4bK4fd1045b Content-Length:0 7 headers, 0 lines Sip read: SIP/2.0 403 Forbidden

I have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong? nxdasterisk-2*CLI> [Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380

Hello, I can't find the information on the wiki :( When is the quota recalculated after a mail deletion ? For instance, I am running low of storage and I use Thunderbird to delete large mail. I only notice a recalculation when I quit Thunderbirdb and I relaunch it. Even, with doveadm CLI, as long as Thunderbird is not disconnected on the client side, the server didn't recalculate the

<P>I write the email again, cause the first one I have had problems while sending it. Here is the email again:</P> <P>Hi everybody,</P> <P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P> <P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled

Skipped content of type multipart/alternative-------------- next part -------------- Nov 21 14:17:47 VERBOSE[32580] logger.c: <-- SIP read from 172.25.103.222:5060: INVITE sip:1715;phone-context=exp_net.ascom@ascom.be:5060;maddr=172.25.96.48;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From:

<P>Hi everybody</P> <P>I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me! </P> <P>1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that when I use g723 I can call MGCP but I can't

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

All, I am connecting to a CS 1000 nortel PBX. I can call out, I have limited success with call in. I get debug traffic that a call is coming in but I get the message "Unable to create/find channel". I was expecting that incoming calls over the trunk would be handled from my sip definition and goto the nortel context. It is not. Below is the actual incoming call debug information. I am

The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not work unless I add "insecure=very" to my "Outgoing settings", but I don't want to do that. I do want to authenticate. Outgoing (Asterisk PBX to Class 5 switch) calls do authenticate and work. The Nortel CS 1500 I'm using as the PSTN-side of my SIP trunk has a username and password that it's

Greetings to All! Help to solve a problem: There is an asterisk and two VoIP a sluice (NSGate 800 2FXS 2FXO, NSGate 800 2FXS). In sip.conf they are registered so: [3301] type=friend host=172.222.135.11 username=3301 secret=0000 defaultip=172.222.135.11 dtmfmode=rfc2833 context=it callerid="VoIPGate2Line1" <3301> allow=g723.1 [3302] type=friend host=172.222.135.11 username=3302

Hello, I am working with 1.8.32.2 which I have patched with t38-gateway and PRACK. t38 is tested and working fine with Zoiper client but I can't get the t.38 software from Biscom (FAXCOM) to talk. In my first attempts I found FAXCOM announces that it supports 100rel so I added the PRACK patch hoping that would do the trick. Now it gets a little further but * complains about rejecting a

I am attempting to get a planet VIP-150T to register with asterisk 1.2.4. After searching google I've found what appear to be instructions in German, Russian and Spanish. Has anyone perhaps seen this before? Asterisk is kicking back the following error: Feb 14 09:59:32 NOTICE[21765]: chan_sip.c:10851 handle_request_register: Registration from '<sip:101@192.168.100.240>'

I'm trying to debug a NAT issue and I can't make up my mind if the problem is with my Vigor 2800 or Asterisk 1.6.2. I know the Draytek is alleged to suffer from nat 'issues' but I did not have the issue with 1.6.1 - so I'm wondering if something has changed? The Draytek offers 'NAT & Routed' on a single device - so my Asterisk sits on a Public IP, and I have a

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 secret=pwd qualify=200 nat=yes host=dynamic