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Voxalot Support Support for the Voxalot service. |
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09-16-2007, 03:56 PM | #1 |
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Join Date: Mar 2006
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Does anyone have Incoming Calls working with a Voxbone, LES.NET or CallCentric DID?
Hi
I've tested with 2 SPA1001s and 1 PAP2. One of the SPA1001 are even on a different home network. All are located in the DMZ. The result of any calls coming in via these DIDs is ONE-WAY audio. All other DID provider numbers work fine. My ATAs are all registered to us.voxalot.com. I have tried making a few changes to NAT settings, but no luck. If anyone has DIDs working from these number. Either your own dedicated DID or through a SIPBroker gateway DID for these DID providers, would you kindly share your ATA configuration details? (or solution) Thanks in advance Richard |
09-17-2007, 02:32 PM | #2 |
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Join Date: May 2007
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Hi
I had no problems with les.net, using either SIPBroker gateway DIDs and *010xxxxxx or my own DID at les.net. My DID is sip-forwarded to xxxxxx@voxalot.com in the les.net trunk setup, the ATA is a PAP2T registered to us.voxalot.com. I will have a look at its configuration when I come home ... Marcel |
09-18-2007, 02:37 AM | #3 |
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Thanks Baikal. Hopefully your configuration will help sort this out. Richard
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09-19-2007, 03:45 AM | #4 |
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Below my settings (SIP & Line1 tab of the PAP2t configuration page, les.net trunk setup). Which way is your one-way audio? I recently observed one-way audio (no outgoing audio!) with another provider which is registered in my voxalot account.
PAP2T-SIP: Code:
SIP Parameters Max [70 ] Max [5 ] Forward: Redirection: Max Auth: [2 ] SIP User [$VERSION ] Agent Name: SIP Server [$VERSION ] SIP Reg User [ ] Name: Agent Name: SIP Accept [ ] DTMF Relay [application/dtmf-relay] Language: MIME Type: Hook Flash [application/hook-flash] Remove Last [no ] MIME Type: Reg: Use Escape Compact [no ] Display [no ] Header: Name: SIP Timer Values (sec) SIP T1: [.5 ] SIP T2: [4 ] SIP T4: [5 ] SIP Timer B: [32 ] SIP Timer [32 ] SIP Timer H: [32 ] F: SIP Timer [32 ] SIP Timer J: [32 ] D: INVITE [240 ] ReINVITE [30 ] Expires: Expires: Reg Min [1 ] Reg Max [7200 ] Expires: Expires: Reg Retry [30 ] Reg Retry [1200 ] Intvl: Long Intvl: Response Status Code Handling SIT1 RSC: [ ] SIT2 RSC: [ ] SIT3 RSC: [ ] SIT4 RSC: [ ] Try Backup [ ] Retry Reg [ ] RSC: RSC: RTP Parameters RTP Port [16384 ] RTP Port [16482 ] Min: Max: RTP Packet [0.030 ] Max RTP ICMP [0 ] Size: Err: RTCP Tx [0 ] No UDP [no ] Interval: Checksum: Stats In [no ] BYE: SDP Payload Types NSE AVT Dynamic Dynamic [100 ] Payload: [101 ] Payload: INFOREQ G726r16 Dynamic [ ] Dynamic [98 ] Payload: Payload: G726r24 G726r40 Dynamic [97 ] Dynamic [96 ] Payload: Payload: G729b NSE Codec Dynamic [99 ] Name: [NSE ] Payload: AVT Codec [telephone-event] G711u Codec [PCMU ] Name: Name: G711a G726r16 Codec [PCMA ] Codec Name: [G726-16 ] Name: G726r24 G726r32 Codec [G726-24 ] Codec Name: [G726-32 ] Name: G726r40 G729a Codec Codec [G726-40 ] Name: [G729a ] Name: G729b G723 Codec Codec [G729ab ] Name: [G723 ] Name: NAT Support Parameters Handle VIA [no ] Handle VIA [no ] received: rport: Insert VIA [no ] Insert VIA [no ] received: rport: Substitute [yes] Send Resp To [no ] VIA Addr: Src Port: STUN [yes] STUN Test [no ] Enable: Enable: STUN [stun.voxalot.com.au] EXT IP: [ ] Server: EXT RTP [ ] NAT Keep [15 ] Port Min: Alive Intvl: Code:
Streaming Audio Server (SAS) SAS Enable: [no ] SAS DLG Refresh [30 ] Intvl: SAS Inbound [ ] RTP Sink: NAT Settings NAT Mapping [yes] NAT Keep Alive [yes] Enable: Enable: NAT Keep Alive [$NOTIFY ] NAT Keep Alive [$PROXY ] Msg: Dest: Network Settings SIP TOS/ Network Jitter DiffServ [0x68 ] Level: [high ] Value: RTP TOS/ Jitter Buffer DiffServ [0xb8 ] Adjustment: [up and down] Value: SIP Settings SIP Port: [5060 ] SIP 100REL [no ] Enable: EXT SIP Port: [ ] Auth [yes] Resync-Reboot: SIP [ ] SIP [no ] Proxy-Require: Remote-Party-ID: SIP GUID: [no ] SIP Debug [1-line ] Option: RTP Log Intvl: [0 ] Restrict Source [no ] IP: Referor Bye [4 ] Refer Target Bye [0 ] Delay: Delay: Referee Bye [0 ] Refer-To Target [no ] Delay: Contact: Sticky 183: [no ] Call Feature Settings Blind Attn-Xfer [no ] MOH Server: [ ] Enable: Xfer When [yes] Conference [ ] Hangup Conf: Bridge URL: Conference [3 ] Bridge Ports: Proxy and Registration Proxy: [us.voxalot.com ] Use Outbound [no ] Proxy: Outbound [ ] Use OB Proxy In [no ] Proxy: Dialog: Register: [yes] Make Call [no ] Without Reg: Register [600 ] Ans Call Without [yes] Expires: Reg: Use DNS SRV: [yes] DNS SRV Auto [yes] Prefix: Proxy Fallback [600 ] Proxy Redundancy [Based on SRV Port] Intvl: Method: Voice Mail [us.voxalot.com ] Server: Subscriber Information Display Name: [ABCD ] User ID: [nnnnnn ] Password: [************* ] Use Auth ID: [yes] Auth ID: [nnnnnn ] Mini [ ] Certificate: SRTP Private [ ] Key: Supplementary Service Subscription Call Waiting [yes] Block CID Serv: [yes] Serv: Block ANC [yes] Dist Ring Serv: [yes] Serv: Cfwd All Serv: [yes] Cfwd Busy Serv: [yes] Cfwd No Ans [yes] Cfwd Sel Serv: [yes] Serv: Cfwd Last [yes] Block Last Serv: [yes] Serv: Accept Last [yes] DND Serv: [yes] Serv: CID Serv: [yes] CWCID Serv: [yes] Call Return [yes] Call Back Serv: [yes] Serv: Three Way Call [yes] Three Way Conf [yes] Serv: Serv: Attn Transfer [yes] Unattn Transfer [yes] Serv: Serv: MWI Serv: [yes] VMWI Serv: [yes] Speed Dial [yes] Secure Call [yes] Serv: Serv: Referral Serv: [yes] Feature Dial [yes] Serv: Service Announcement [no ] Serv: Audio Configuration Preferred [G729a ] Silence Supp [no ] Codec: Enable: Use Pref Codec [no ] Silence [medium] Only: Threshold: G729a Enable: [yes] Echo Canc [yes] Enable: G723 Enable: [yes] Echo Canc Adapt [yes] Enable: G726-16 [yes] Echo Supp [yes] Enable: Enable: G726-24 [yes] FAX CED Detect [yes] Enable: Enable: G726-32 [yes] FAX CNG Detect [yes] Enable: Enable: G726-40 [yes] FAX Passthru [G711u] Enable: Codec: FAX Codec [yes] FAX Passthru [NSE ] Symmetric: Method: DTMF Tx [Auto ] FAX Process NSE: [yes] Method: Hook Flash Tx [None] FAX Disable [no ] Method: ECAN: Release Unused [yes] Codec: Dial Plan Dial Plan: [(x.|x*x.|**x.|*x.|<#:*>xxx.) ] Enable IP [no ] Emergency [ ] Dialing: Number: FXS Port Polarity Configuration Idle Polarity: [Forward] Caller Conn [Forward] Polarity: Callee Conn [Forward] Polarity: Line Enable: [yes] Code:
Peer Name NNNNNNNNNNNNNNN Your Description [forward to voxalot] [?]Peer Technology [SIP ] [?]DTMF Mode [rfc2833] [?]Re-Invite [no ] [?]NAT [no ] [?]Error Method [Verbal] [?]Codecs[*]G.711 [*]G.729 [*]GSM [*]G.726 [?]Peer Type [URI ] (URI/PSTN Experimental) [?]Peer Address [nnnnnn@voxalot.com ] [?]Password [ ] [?]Registered No [?]Registered IP nnnnnn@voxalot.com [?]Registration Expires No [?]Outbound CallerID [ ] [?]7-Digit Dialing [ ] [?]7-Digit Area Code Prefix [ ] [?]10-Digit Dialing, Prefix 1 [ ] |
09-20-2007, 12:06 AM | #5 |
Join Date: Apr 2006
Location: Vancouver, BC
Posts: 296
Thanks: 94 Thanked 53 Times in 27 Posts |
My Voxbone DID (which is mapped to my Voxalot SIP URI) works fine, two-way audio.
NAT Support Parameters Handle VIA [no ] Handle VIA [no ] received: rport: Insert VIA [no ] Insert VIA [no ] received: rport: Substitute [yes] Send Resp To [no ] VIA Addr: Src Port: STUN [yes] STUN Test [no ] Enable: Enable: STUN [stun.voxalot.com.au] EXT IP: [ ] Server: EXT RTP [ ] NAT Keep [15 ] Port Min: Alive Intvl: I recommend changing "Send Resp To" to Yes. Normally it should be Yes if "Substitute VIA Addr" is Yes. That might help. If not try changing the other four VIA settings to Yes and those two back to No (either the top four should be yes, or the fifth and sixth ones should be yes, not both at once--I am told). If still no luck, DMZ your PAP. |
09-20-2007, 04:22 AM | #6 |
Senior Member
Join Date: Mar 2006
Posts: 153
Thanks: 37 Thanked 7 Times in 7 Posts |
Thanks Baikal and Ctaylor.
I'll try playing with the setting again tomorrow and let you know. BTW, my device is in the DMZ already. Take care Richard |
09-20-2007, 02:00 PM | #7 |
Join Date: Apr 2006
Location: Vancouver, BC
Posts: 296
Thanks: 94 Thanked 53 Times in 27 Posts |
I have the weird relationship of putting my ATA in the DMZ of my router, while keeping the various NAT compensating/port-busting techniques of the ATA enabled (in my case the fifth and six options to Yes), actually kills my device and cuts out voices. Just tried again now, and still have that happen, so I have to leave off the DMZ mode unless I want to investigate further what the settings should be.
I recommend playing with the STUN settings while not having any dmz or special port forwarding rules enabled for your ATA. |
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