similar to: Asterisk behind NAT, Polycom behind NAT (SIP), how to work?

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 List; Any one can advise me to a good link to see and buy Polycom IP Phones? Also, if I need support (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. Regards Bilal ____________________________________________________________________________________ Be a better

Hi List; I heared that IAX is good for NATing issues, but I do not know if it can help me in that senario: I have two Asterisks machines in different sites and both are behind NAT (both have private IP address), I need to link these two asterisks with IAX trunk (if it help really in such senario), but I do not know if it will work without doing special routing settings on the router (like

Hi All; I was using since one year Asterisk 1.4.19.2 and zaptel 1.4.10.1 and they were working fine via SIP, IAX and Digium fxo and fxs ports. Suddenly just before 2 or 3 days, the voice become garbage like robot when I place a call from the SIP Phone (which is in a country and the Asterisk box in another country). I am surprise what is the reason that let rtp become like this ! The sound now

Hi All; Anyone knows an IAX IP Phone works fine and tested? Does polycom support IAX IP Phone? Regards Bilal

Dears; I am facing a problem in disconnecting the calls, it is related to the rtptimeout (disconnecting if there is no RTP packets from both sides). My Asterisk Box is behind NAT but there is a static real IP address at the ADSL router. We call from the Mobile to the PSTN analogue numbers which are connected to Asterisk Analogue card (the telephone lines are analoge), and then we dial the

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: "It shouldn't matter as long as you are handling the NAT correctly your end. We do not fix NAT so if you're sending internal addresses in your INVITEs or SDP then things will fail but if you're handling it correctly, we shouldn't tell the

Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards Bilal

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

I am having an issue with doing FTP auto provisioning of Polycom 501's when they are behind a NAT. If I put the phone on the same subnet as the provision server it loads the configs and changes fine but as soon as I put in behind a NAT it comes up with cannot contact boot server. I have tried behind and replicated this behind a PIX 501, a Linksys SOHO router and a Motorolla SOHO router.

Hi List; How can one Asterisk Box A to send a SIP call for another Asterisk Box B, and that call to be authorized based on the username and password, and not on the IP (as the IP address of the source is not known because it keep changing)? I think the trick in the Dial command, how to write it properly in a way that other Asterisk Box can recognize the sip username and password which are existed

Dears; I do not know if any had experience in using speex or ilbc with IAX and got good results, because I am facing a problem with GSM. I am facing a noise problem when I am using GSM with IAX trunk as following: IP Phone (G711) ---> Local Asterisk Box ---> IAX Trunk using GSM codec ---> Remote Asterisk Box ---> Digium Card (FXO) to terminate the call to the destination While no

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Hi List; Can someone advise me which IP Phone model that has buttons that can be assigned to do specific functionalities (call pickup, call formward, call appearance) and a transfer button and hold button? Which is the best of the following (that has buttons can be assigned to specific functions): Cisco 7970 or 7960 Polycom 501 Grandsream IP Phone Budge Tone 1001 or 1002 Linksys SPA 942 or 922

Hi All, I have polycom phones setup in an office connected to a cloud asterisk server. The polycom phones can call out just fine - audio just fine. However a call coming into the cloud asterisk answers fine - get the autoattendant, enter the extension and the polycom does not ring. The CLI shows that the correct SIP extension is being Dialed (SIP/524) Looks like I'm getting a 401 permission

Hello; Anyone used PoE L2 network switches other than cisco and recommend this for us? We need it to be stable and costly effective. Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130714/54feca5a/attachment.htm>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Is it possible to use the instant messaging feature of Polycom phones in Asterisk? At the moment I'm seeing this in the SIP messaging when I try to send one from a Polycom 450. <--- SIP read from UDP:<CENSORED POLYCOM IP>:5060 ---> INVITE sip:0100@<CENSORED>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <CENSORED POLYCOM

Hi, Has anyone used polycom phone IP500/600's conference feature? I ran into some issues. Polycom is not behind NAT. Polycom calls pstn number A, (gets two way voice), presses conference, (A is put on hold), calls pstn number B, (gets two way voice), presses conference again, suppose to have all three parties in the call, polycom and B have two way voice, but polycom doesn't have two

Hello All; Anyone can advise for the good phone (Polycom, Linksys, ... etc) that is a stable and support the codecs: g723, g729, and speex? Actually I would like to have the speex codec because it have the ability to compress to very high compression so we can work with the low bandwidth (for speed about 3 or 4 kbps). I tried Grandstream but really it is a bad device and not worthy to buy it or