similar to: Polycom and forwarding.

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

Somebody was helpful enough to give me the very latest release of Polycom's firmware (2.1.0). Unfortunately, I still get that issue. So I'm stuck asking again: Anybody ever got that? Mike _____ From: Mike [mailto:list@virtutel.ca] Sent: Wednesday, April 11, 2007 13:37 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Polycom 501 issue with latest

Hi, I've had a functioning Asterisk system (1.2.18), which I haven't reconfigured in any way, that is just now refusing to forward calls. I only have Polycom phones. When I use the phone's forward feature (forwarding the phone with extension 204 to extension 206, which used to work as recently as yesterday) I get this in the console: "called sipreg-12344". No ringing,

Hello List, Since I'm looking for a new VoIP provider for US origination/termination, I will very appreciate if you can chare your experience with Flowroute, Vitelity and Voip.ms Thanks in advance! Elder D. Arohuanca dCAP 1497 Lima - Peru -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130617/c0e347d9/attachment.htm>

Hi, all. I want to have a button on my receptionist's 601 that, when pressed, will forward her current call to a given extension. Is there any way to do that? I've tried to RTFM, but I'm coming up empty. Thanks, -Ken D'Ambrosio

Hi, I've upgraded a few Polycom 501 to SIP 2.0.3b (can't get 2.1.0 because of Polycom's firmware policy, but this is the "latest publicly available" from Polycom's web site). I've noticed that some keys get "sticky" though. Soft buttons for example (i.e. "end call") need to be pressed 2-3 times for them to react. I've downgraded to

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667)

If I do a supervised forward on a call (Polycom 501, Asterisk 1.2.1), all goes well until the second time I hit forward (to join the caller with the extension); then, the caller's MoH goes away (making them think they've been hung up on), and the server spits out: asterisk-cw*CLI> <-- SIP read from 10.20.2.16:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP

Hello. Yes, i think that was the cause why agent forwarding wasn't performed at all, i had to rm(1) the control socket and the next ssh(1) connection forwarded the agent normally again. (It was a huge timeshift by several hours.) I.e., just in case this is something you didn't have on your radar yet. Ciao. --steffen

Trying to figure out the best way to pull an active call out of a queue by unique id and put it on hold. I don't want to put it on hold on the agent's phone but I want it to be pulled away from the agent's phone and into Asterisk limbo somewhere. Shortly after I want to pull the same call out of limbo and redirect it back to either the same agent or another. I was thinking about call

Hi, all. I see that the Polycom SoundPoint IP 330 supports VLAN... but I don't quite see how that works. Do you point a non-VLAN'd segment at it (akin to when you uplink a VLAN_enabled switch), and have the phone implement the VLAN? Or...? *puzzled* Thanks much, -Ken

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

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Hi guys, I am trying to make a trunk between two asterisk system SIP Trunk on Asterisk 1.6 but I cannt make it work, can any body help me plz? Thank you

Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance.

Hello, I need to setup system of aroud 60 DECT phones with asterisk. So far I found http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710 However is there some cheap base station(similar to GSM cell) so I can handle all DECT phones centralized and plug it inside asterisk ? Thanks, Peter

We have a new customer with a lot of old phones like the 9133i. They won't register, and we see some very strange behavior with them. If the SIP peer exists, they simply fail silently, with no error in the CLI or the messages log. Nothing works, but no errors. If the peer does not exist, it's clear that it's registering improperly: [2013-04-28 13:34:31] NOTICE[3058] chan_sip.c:

Docs? Polycom has docs? Where would one find this fabled land of... err I mean Polycom does stating what ftp servers are supported? Doug. -----Original Message----- From: Ken D'Ambrosio [mailto:ken@jots.org] Sent: Tuesday, March 07, 2006 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: