similar to: SIP transfer issue

Hello, I have Elmeg IP290 phone and have problems with VM. I don't know how to configure this IP phone, that it could call to *97@192.168.0.1 if I pressed "VMail" button. Now if I press buttom "VMail" , ip phone dials: sip:asterisk@192.168.0.1 (192.168.0.1 - Asterisk IP). So I don't understand , from where it takes "asterisk", cause I have never write

Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID

I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console. Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '16749440-c28be02e-64b73be7@172.31.16.67'. Both legs must reside on Asterisk box

Am 14.06.2020 um 17:33 schrieb Luca Bertoncello: Hi So, I got a phone (Elmeg IP290) from a collegue and tested it... > What I'll do tomorrow with a test phone is: > > 1) connecting it to my Asterisk and try to make a call > 2) connecting it directly to the servers of Deutsche Telekom (using my > network) and try to make a call Absolutly *no changes* on the behaviour

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. Here's the REFER that the phone at 2944093 sends directly to Asterisk: U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER sip:3254102@216.186.142.203 SIP/2.0. Via: SIP/2.0/UDP 216.186.128.68;branch=z9hG4bKba3b074892377BD1. From:

I just realised my problem seems to be related to bug 0003710 - "0003710: [patch] Consultative transfers between asterisk servers". It's unclear from the bug info if this problem has been resolved yet. Anyone know? Doug. > -----Original Message----- > From: Douglas Garstang > Sent: Monday, March 27, 2006 4:41 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Has anyone gotten around the general problem where you have multiple Asterisk servers in a cluster, any of which may take a call. If you transfer a call from one Asterisk system to another, the second has no idea of the first call, and the first refuses to release the call and logs: May 5 12:40:49 NOTICE[2864]: chan_sip.c:6758 get_refer_info: Supervised transfer requested, but unable to find

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. It appears this related to bug 3710. It's unclear from the bug if the problem has been fixed or not. If it hasn't, then this seems pretty serious and would I guess affect any NAT-ted phones ability to transfer calls. Here's the REFER that the phone

Greetings list, Wondering if anyone's come across this before. I've configured a couple of our servers with a "privatedundi" context to allow calls to still flow between extensions even if they're registered to different servers . The DUNDi lookups seem to work fine, evidenced by the following on the originating server: -- Called

Mine is very similar: i don't have echocancelwhenbridged=yes because it seems work only on TDM, is it ? And in Italy, I often have set pridialplan = unknown About echo I have some problems, but only at the beginning of the call. After 3-4 seconds the echo became almost null, specially with snom 190; with pa168s and ywh10 I have again some problem, the echo come up also after 1 minute of

Hi! I have a little problem with the given phone... Do someone know it? My problem is that I'd like to display the name of the caller (if it is saved in the address book, of course), but it always display just the number... Thanks Luca Bertoncello (lucabert at lucabert.de)

Greetings list, I posted this to the -biz list a few days ago. In hindsight, I think it would have been more appropriate posted here, so apologies to those on both lists who've now seen this twice. I have had a request to provide 2x PRIs to a site in Lausanne, Switzerland, but my knowledge of the Swiss Telco market is non-existent. Are there any folks on the list who've experience in

Greetings list, Not exclusively asterisk-related, but I've noticed the CentPBX site has been offline the last few days. Anyone know the reasoning behind that, and more importantly, is anyone mirroring it? Thanks in advance. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it This email is made from 100% recycled electrons

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with

Greetings list, Wondering if some kind soul can help me with an issue with chan_sip segfaulting as soon as it loads... Basically, if sip.conf contains any peers with "host=dynamic" in them, asterisk won't start. Doing -vvvdddc yields the following: [chan_sip.so] => (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found Segmentation fault As

Hi, We are working on a product to analyze pcap files of VoIP calls. So far it does a reasonable job of analyzing the frequency distribution of packets in both directions, pointing out which direction packet loss / bad jitter occurs.  If you can trap the traffic on the outside and the inside of your Banana Pi and send me the pcap files, I would be happy to run it through our analyzer as

Greetings list, I need to set up a point to point SIP connection between two devices without either of them registering with a registrar/proxy/etc. at all. The devices I've tested so far all seem to insist on having a registration before they'll make or take calls. One of the devices needs to be an ATA with an FXO port (e.g. Sipura/Linksys SPA-3000/3102), the other device can be either

Hi list! While looking for the Snom 190 I found another phone, the Elmeg IP 290 (www.elmeg.de). Looking at the pictures & the specs they seem to be very similar beasts but the firmware is supposedly not interchangeable. Does anyone know the difference between the 2, do they work with Asterisk? The weird thing is that Elmeg has similar phones with the Snom look but they are ISDN only (no