similar to: Problem with call transfer and Polycom 430

Hi all. When a user transfers a call by pressing the "transfer" soft button on their phone, I'd like it to "beep" at them when the transfer is complete. I've got it turned on in features.conf: xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer However, it seems that

Hi all, Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device to automatically answer? I can do this with my Polycom phones and would like to do it with my ATA's. Any ideas? -- Take care and have fun, Mike Diehl.

Hi all, I seem to be having a strange problem with a sip trunk. On a fairly frequent basis, I'll make a call, ore receive a call, and there will be NO sound. The strange part is that both endpoints are public IP addresses so NAT isn't in play and a sniffer trace reveals that the packets simply aren't being sent. It only seems to happen on a particular trunk. The same phone

Hi all, I've got 12 Polycom 430's behind a NAT that are working pretty well except for one thing: If one of my users is on the phone when another call comes in, there is about a 10 second time during which the user can't hear the call they're on. Then it returns to normal. My research indicated that I needed to add these setting to the phone:

Hi all, I'm trying to use a wrt54gl router running dd-wrt as a provisioning server for a remote installation. I've got dhcp working and I have provisioning files ready to go. I understand that I need to set bootp option 66 to point to the tftp/ftp/http server. In fact, I have this working completely with the ISC dhcp server The problem is that I don't know how to get

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

Hi all, I used to configure each of my sip clients with a unique identifier via setvar. These clients were all configured as "friends." However, now that I've got some Polycom phones, which MUST be "peers," I am unable to define this variable. For example, this works: [friend-client] context = default accountcode = pcc type = friend username = username secret =

Hi all. I'm in the process of configuring a phone system for my family and friends. I'm wondering if I should try to implement the "Vertical Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the Asterisk dialplan, or if I should delegate those functions to the various ATA's. For example, the Sipura SPA 2002 can handle*69 internally. On the other

Hi all, I've got a batch of Polycom 335's that I'm trying to get setup. The phone works fine, but when I plug a PC into the PC port on the back, the PC can't get to the Internet. I've turned off all of the VLAN configuration. I've never had this problem before, so I'm at a loss as to how to proceed. Usually, it just works... Any ideas? -- Take care and

Hi all- Equipment: Xlite softphone Polycom 301 with SIP 2.1.1 and BootROM 3.2.3 Polycom 501 with SIP 2.1.1 and BootROM 3.2.3 Asterisk 1.4.2 SIP Trunk to FWD I wanted to post this problem as I haven't found it described in any of the past presence threads on here. I use an identical configs for a Polycom 501 and 301. (I actually unplug one when the other is in use). The

We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

Hi, I have a really strange problem with a Polycom 430 phone and Asterisk 1.4.20. Currently If I dial the Polycom from my mobile phone answer the call on the Polycom and then hangup the mobile the call ends fine on the Polycom. But if I call from the Polycom to my mobile and then I hang up the mobile the Polycom thinks the call is still active. However doing a show sip channels shows the the

Hi List; Can someone advise me if Polycom support H323 that work fine with Asterisk? And wether this H323 Polcyom devices more costly than SIP Polycom. Also, I am not able to know if new Polycom come with PoE adaptor so no need for PoE Switch (can use normal switch that does not support PoE)? Do I need any special cable for Polycom or normal Ethernet cable? Regards Bilal

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

Hi all, I've got a Polycom 501 that's been working with Asterisk for some time. However, I don't seem to be able to put a call on hold and get it back. It goes on hold just fine. But when I press the resume button, nothing happends. Anyone seen this befor? Any ideas on where to start to fix it? TIA, -- Take care and have fun, Mike Diehl.

I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on a flat local network. I followed the provisioning guides that I found on the Web, and I have the phone downloading bootrom.ld, sip.ld, and a bunch of configuration files. This all works properly. However, I receive the following error: NOTICE[27345]: chan_sip.c:14725 handle_request_register: Registration from

Hi all, Lately, I've been seeing more and more instances where I get a flood of warning messages like this: [Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining The next thing I know, my server is dropping calls and starting to misbehave. I use fax via T.38, so I can't just turn udptl off. I could expand the port range, but I suspect that will just mask the situation.

The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't know where to look. p From: "Von L." <methodvon@gmail.com> To:

Hi, The Polycom 600 phones do not natively bridge with Asterisk. I've solved the problem, but I'm not sure how general it is, so I thought I'd ask this list for advice. It's necessary to use a recent Asterisk CVS for this, since there was a problem with session versions in earlier CVS builds. The problem now is the Via field. When the reinvite goes out, the branch number