similar to: SIP subscribe with multi-server registration

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000

I'm currently working with Asterisk 11.8.1 trying to get Multicast RTP working (it's not) with some Polycom phones, and I'm really trying to determine if Asterisk or the phones are the issue. I THINK it's Asterisk... In extensions.conf I have a simple: "Page(MulticastRTP/basic/x.x.x.x:xxxx) line, and when I dial that extension I get: -- Called

Hello, I am using OpenSIPS to register all the users and planning to use asterisk for Auto Attendant, Queues, Voicemail and Conference Bridge. I have a scenario where the signaling does not happen properly: 1) A user from Opensips dials an extension 7000 which is an auto-attendant extension. The call is routed to asterisk to play the auto attendant messages like Welcome and Dial the

Hello, Has anyone on the list been able to successfully setup a Polycom Soundpoint 500 IP phone? I am getting failed registrations, and the Polycom documentation is not very precise. Their web interface isn't helping much either. Thanks in advance, Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

We have a system where calls are coming in from telcos via an opensips server and then being redirected out to a regular sip destination. There is no NAT, DTMF features, call recording, or codec translation being performed so I would expect asterisk to issue a reinvite after the call is answered and switch the audio however it is not happening. Here is the sip peer information for the call

Hi Everyone, Configuring a Polycom conference bridge IP 5000 to connect to Asterisk. For some reason I don't see any SIP packets coming in to Asterisk at all. I don't want to use XML or ftp etc for now and just use the Web Interface to get it going with basic features. But the Web UI is a bit confusing with SIP and Line tabs. I have put this on the web interface: SIP > Outbound

I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. > Thanks Olle, > > So am I to understand that you

Hello All, I have a little complicated question about the Dial command. I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on Asterisk servers. Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server. Everything works except for trunk numbers: For each peer on Asterisk, "Addr->IP" is IP of the Proxy and "Reg. Contact" is the IP

I have 2 Polycom SP 500's attached to my system. Both are behind NATs, but both seem to work fine, for the most part. A few weeks ago, I started to notice that I get an error message from one of them: Feb 20 08:54:58 NOTICE[10663]: chan_sip.c:7691 handle_request: Registration from '<sip:Polycom1@xxx.yyy.93.254>' failed for 'zzz.aaa.103.75' However, trying to call the

trying to set up and configure a polycom soundpoint ip 500 phone, when trying to get it to register with sip, i get the following message Sip read: REGISTER sip:67.110.252.13:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58 From: "138polycom" <sip:5622832456@67.110.252.13:5060>;tag=B8D9FA39-9D85A6AC To:

Hello List, I have been trying to configure a sip profile ( peer / friend ) for each of my customers behind a sip proxy for some time, but I have had no success, so I would appreciate your help. Customer -> OpenSIPS -> Asterisk -> PSTN The opensips is working as a sip proxy with record route, for billing, load balancing and authentication purposes. I would like to be able to define

We're having an odd issue with codec negotiation from one of our SIP providers. Here's the basic situation. We receive an invite from them advertising support for G711, G729, and G723. In our response, we send back that we support G711 and G729. In about half the cases, this results in no problems, with audio being encoded with G711. The other half of the time, they send us a second

Somewhat off topic. We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6 The phone will no longer send SIP subscription messages for buddies to Asterisk. I have broken the directory file down to make it as simple as possible. Here is what it contains. <?xml version="1.0" encoding="UTF-8" standalone="yes"?> <!-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05

Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them? I called Polycom tech support, who where utterly useless. Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and

When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from '<sip:201@192.168.2.13>' failed for '192.168.3.2' - Not a local SIP domain In sip.conf I have this for my global settings: [general] context=from-sip ; Default context for incoming calls

Hello; We were able to set up shared folders in a cluster (using dovecot as proxy on dedicated front-end servers, without using director) by following https://doc.dovecot.org/configuration_manual/shared_mailboxes/cluster_setup/. Here is our related conf on imap farms (dovecot v2.3.16 on Almalinux): namespaces: mail_location = maildir:~/Maildir namespace default { ? inbox = yes ? location

> On 02/03/2023 16:21 EET Julien Nadal <julien+dovecotnl at mujik.fr> wrote: > > > Hello; > We were able to set up shared folders in a cluster (using dovecot as proxy on dedicated front-end servers, without using director) by following https://doc.dovecot.org/configuration_manual/shared_mailboxes/cluster_setup/. > > Here is our related conf on imap farms (dovecot

Hello list, I was wondering if any of you guys have had any luck with polycom in remote offices, I'm facing a weird issue, polycom phones work fine in the main office, in remote office it says, Registration from '<sip:202@10.0.1.190>' failed for '70.59.21.112' - Wrong password the odd thing is Linksys phone works without any issue!! polycom wont register but its able to