similar to: Call Hold problem

Hello All, I have Cisco 7960's, Cisco 2950 Switch. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone is disconnects at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone seen this? Thanks for the help,

Quick update on my issues, Voicemail doesn't pickup also. It just drops the line.. Thank you Chris Tuska ------------------------------ Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the

Hello All, I have Cisco 7960's, Cisco 2950 Switch, SUSE 9.2, PIX Firewall. Here is my issue I can dial out no issues but when someone calls in the phone rings I answer and the phone disconnects the call. Call from my cell to my house I answer the cisco phone it then disconnects the call at the same time on the cell I hear 4 beeps and about 5 secs later the line on the cell drops, as anyone

Hello all, I am running * 1.2.6 I have 2 linksys PAP2 with two phones each. Until recently all was good. on Friday I was running 1.2.5 when I added the fourth phone. I have to admit to initially wiring the rj11(crossed wires) wrong the first time but other than that nothing I can think of. Added the appropriate entries in sip.con and on the PAP2. I then tried to call from one line to the

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in

Hello everybody and merry xmas. I have a problem with sip phones calling each other inside the same network (no nat, no firewall). You can make and receive calls and pick them up, but you cannot hear anything on any side of the call. But if you press hold/unhold or you transfer the call, then everything works as expected. Ths SIP phones I've tried are Swissvoice IP10s and kphone, it

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hello list, i need help. My problem is that when I want to call (using ISDN phone or internal SIP client) via the Sip provider a real phone number (ISDN phone or internal SIP Asterisk >> SIP ), I get a ring tone. When I now decide to hang up (e.g. if nobody answers), the called telephone continues to ring almost forever. the sip.conf: [2563105] accountcode = 2563105 username =

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

I'm trying to integrate my Asterisk box with my call manager 8 server. I can call from the call manager to a phone on asterisk, but I can't call from a phone on asterisk to call manager. Any help would be greatly appreciated. sip.conf [2000] type=friend secret= dtmfmode=rfc2833 host=dynamic canreinvite=no context=myphones allow=ulaw nat=yes [2001] type=friend secret= dtmfmode=rfc2833

Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan <hvardan71 at gmail.com> Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To:

On Fri, 26 Jun 2015, Ludovic Gasc wrote: > 1. What's the "official" notation of each line: "=>" or "=" ? In the > wiki of Asterisk, I see very often "=>", however, what's the reason for > both syntaxes authorized ? Historical ? I'm not 'official,' but I have a strong preference for just '=.' Using

Trying this again after my first away from work in a couple weeks. Running Asterisk 13.0.0 IP authentication with Vitelity I can Originate with sip, but not pjsip. Here is the sip settings and trace. Action: Originate ActionID: S8 Channel: SIP/8005555555 at outbound.vitelity.net Exten: createcall Context: TestApp Priority: 1 Timeout: 60000 CallerID: John Doe <1234> Variable:

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)

I'v a sip client and a sip trunk to FWD: [general] port=5060 context=default tos=reliability disallow=all allow=ulaw careinvite=no [freeworlddialup] context=default type=friend username=MYUSERNAME secret=MYPASSWORD host=fwd.pulver.com [igor] type=friend callerid="Me" host=dynamic dtmfmode=rfc2833 careinvite=no When i try to call a FWD number from SIP client i obtain a lot of

Hello, I have been trying to completely disable music on hold on my asterisk system. When a call is put on hold I do not want any music on hold, but I would like the remote user to get informed of this event (depending on the technology e.g. with a SIP reinvite and an SDP indicating the call is on hold). I have searched and tried out various approaches, but when putting the call on hold

I have a DID but for some reason is not working in asterisk-1.6 The same sip connection in asterisk-1.4 is working OK, but it doesn't work with asterisk-1.6 Here is my sip.conf section: ... [actio-out] type=friend secret=password user=48746612254 username=48746612254 fromuser=48746612254 authname=48746612254 callerpage=48746612254 fromdomain=sip.actio.pl host=sip.actio.pl insecure=very