[Thinktel Support Center] Recently Changed Articles

How do I factory reset my Linksys or Cisco phone and reprovision it?

Thu, 17 Jun 2010 13:29:13 GMT

Factory Resetting Phone Select the menu button on your phone (below voicemail button, looks like a sheet of paper with a folded corner) and scroll down to the Factory Reset menu choice. Select this feature and follow the steps to factory reset the device. Once completed and the phone has powered back up to main screen press the options button on your phone and scroll down to the Network menu choice. In here you should find the phones ip address on the local network. Go to your computer, open your web browser, and enter http://192.168.x.x replacing the x.x part with the ip address obtained in the address bar of the browser and navigate to the site. You'll see an administration panel where you can view the status of your phone as well as configure it. At the top right section you should see two links. Click admin first, wait for the page to reload. Next click on advanced and wait for page to reload. Provisioning the phone Click on the Provisioning Tab and ensure 'Provision Enable' is set to yes Set 'Profile Rule' to: http://provision.thinktel.ca/init.cfg Click on 'Submit All Changes' Your Cisco/Linksys device will reboot up to four times with firmware potentially being upgraded if out of date Please note: It can take up to 10 minutes for your device to populate in our provisioning servers Quote your device MAC address or serial number when placing orders for VoIP service on your Cisco/Linksys hardware

Support for the T.38 Protocol

Thu, 06 May 2010 20:09:55 GMT

ThinkTel supports T.38 in the following scenarios: 2-way Direct media VoIP calls. 2-way TDM-to-VoIP (and vice versa) calls. 2-way TDM-to-TDM calls between two different Universal Media Gateways (UMG). T.38 is not supported in the following cases: Non-direct media VoIP calls, regardless of whether T.38 is supported by the endpoints. Multi-way calls (including tapped calls, for instance). Faxes using V.34 Calls where an Originating or Terminating Application Server has been invoked. BLES-to-VoIP (and vice versa) calls. Calls where no leg is over IP (!) Behavoir that initates T.38: 1. If a fax-specific tone is detected, the call will be reprogrammed to G.711 (if it is not already using it). 2. If a reINVITE was received from a SIP device, it will be rejected. This should not cause the call to fail - instead the SIP device should send a reINVITE requesting the use of G.711 instead Note that some tones are used in both fax and modem calls. If such a tone is detected on a call using a low-bandwidth codec, the CFS will reprogram the call to G.711. If a fax-specific tone is subsequently detected, the call will only then be reprogrammed to use T.38. This "double fallback" is actually the normal case, since fax calls tend to start with the CNG or CED tones, and these are also used in modem calls.

Supported routers/firewalls for Thinktel voice traffic

Thu, 06 May 2010 19:10:50 GMT

Problem:

Do I need a special router/firewall for voice traffic?

Solution:

Not all NAT firewall/routers can handle voice traffic properly. Symptoms of this include inability to receive calls inbound, dropped calls and no one being able to hear you when placing an outbound call. Most firewall/router devices have firmware upgrades available to deal with SIP properly, so its always good measure to upgrade to the latest stable firmware. Many older devices have no updated firmware and may need to be replaced.

* Routerboard Series All Routerboard models have excellent support for VoIP services. Great support for tunning QoS rules. ThinkTel uses the Routerboard 750 and other models on all our retail installations.

*
* Linksys WRT54G Hardware version 5 requires new firmware immediately after being removed from its box.Version 4 and older are fine.

* Linksys WRT54C All versions appear to work fine.

* Linksys WRT54GS (non SRX model) All versions appear to work fine.

* AOpen Broadband Routers Work out of the box and are cheap at about $30.00

* Linksys SRX gear (all models thus far) cause us registration problems with Sipura/PAP2 devices. NAT keep alive fails which results in loss of inbound calling.

* D-Link DI-604 and friends have some problems with early NAT timeouts, causing temporary failures (this has been hit or miss)

* Cisco PIX 501 - 525 Version 6.3 firmware and greater works with or without SIP handling enabled.

* Cisco IOS with SIP CBAC disabled

* OpenBSD software firewalls (FreeBSD works fine, OpenBSD loses NAT sessions)

* Windows 2000/2003/XP Firewall/NAT- Not a good thing, introduces extreme jitter into the stream, but works

* Linux IPTables Works fine, sometimes needs higher UDP timeout setting, but not normally

* SPA-2100/SPA-2102 as a router by itself works fine (with other VoIP behind it)

What should my Quality of Service settings be ?

Wed, 24 Mar 2010 19:30:00 GMT
Your site router (or layer 3 switch) should be set to have the DSCP or ToS setup with: SIP TOS/DiffServ Value: 0x68 RTP TOS/DiffServ Value: 0xb8 SIP is for call signaling and RTP is for the media paths on the live call.

Inbound Caller ID is wrong on my Toll Free numbers

Tue, 19 Jan 2010 16:26:08 GMT
Toll-Free Numbers often need the inbound caller ID to terminate the call properly. Often cell carriers will pass the pilot number of their clients local rate centre instead of their actual number. This is the wrong inbound caller ID you are seeing on toll free number. Unfortunately, there is nothing we can do about this.

Can I test my internet link for VoIP?

Tue, 15 Dec 2009 21:40:11 GMT

To test your internet access for voice media, follow this link:
Thinktel VoIP test

I receive no caller ID name on my Rogers Wireless handset

Thu, 04 Jun 2009 19:04:50 GMT
This is a direct issue with the Rogers network and their policy on how they handle caller ID from remote networks. As explained by one of our upstream providers: Calls from analogue lines on the MTSallstream network displays name and number on Rogers cell phone. Calls that are originating from other sources (such as pri/sip etc) which have a different method of generating name display are discarded by Rogers gateway switch. I worked with Rogers and confirmed by Rogers trace that name was being passed along. I also spoke with Bell IXC and they stated that this was a known problem with Rogers. They spent several weeks troubleshooting this with them. Unfortunately, there is nothing more I can do with this issue.

Successful SIP PBX Interops

Thu, 28 May 2009 22:37:19 GMT
- Communigate Pro - Vonexus - Broadsoft - Nortel SIP Gateways - CCM (Cisco Call Manager) - Asterisk 1.2.x - 1.4.x - CallWeaver - Sutus - Broadsoft - Dialogix - Avaya - Toshiba Strata - Fonality - Acme Packet - Siemens - Natural Convergence - Pingtel - Epygi - Linksys - AdTran - VegaStream - Audio Codes - Next Tone - Freeswitch - Cisco UC520 (Call Manager Express & Call Unity Express) - Microsoft Response Point

What is a pilot number used for on my SIP Trunk?

Wed, 20 May 2009 00:26:06 GMT
Every SIP trunk has a number in one of our rate centres that is used to identify you in CDRs from our swich, including defining the local calling area of your SIP trunk. This is a pilot number. It is used in the same way that a telco would deliver a PRI to your location; however the difference with our SIP trunk is that you can add DIDs to it from any of our rate centres, where a local PRI is just DIDs in that local rate centre that is defined by its pilot number.

Freeswitch sample config

Tue, 31 Mar 2009 15:28:38 GMT
By spec, our SIP trunk only authenticates by IP address. For the case of Freeswitch, we can enable authentication in order for an error free configuration. The following the minimal needed to establish a functional SIP Trunk with JumpShop: Realm: vp.jumpshop.com or eico.jumpshop.com Username: given to you. Password: set in the same place Also enter the proxy to be either vp.jumpshop.com or eico.jumpshop.com.

I have an IP phone/ATA that I would like to peer to Thinktel. How do I do that?

Wed, 03 Sep 2008 20:21:38 GMT

These are the essential configuration values for your phone/ata will be:
1. The SIP username. This will be the 10 digit number we provide.
2. The SIP password. This password we will provide for you to register to our switch.
3. The SIP proxy. Set your phone/ata's SIP proxy to be: vp.thinktel.ca (Setting the Outbound SIP Proxy is typically not needed)

These values should allow proper SIP registration and call flow assuming your local network handles with SIP properly.

Is my Linksys/Sipura SPA3102 supported by your company?

Fri, 15 Aug 2008 19:36:06 GMT
We do not support the Linksys/Sipura SPA3102 because of problems with the firmware related to T.38 and provisioning. Please see the following URL for a list of supported devices.

http://wiki.thinktel.ca/Provisioning_with_Thinktel

Grandstream GXE issues

Fri, 15 Aug 2008 19:17:43 GMT
The GXE has a 'SIP Keepalive' setting that needs to be disabled or set to 'none' within the web adminstration of this PBX. It uses a keep alive method our switch does not understand.

SNOM phones cannot receive or make calls on my PPPOE internet connection

Wed, 30 Jul 2008 18:03:56 GMT
SNOM IP phones are known for sending very large packets (1500 bytes) that may be dropped by routers that are connected to PPPOE internet connections. PPPOE internet connections have a maximum packet payload of 1492 bytes, which may not be fragmented by the site router and simply dropped.

A solution to this is to use a router that will fragment the packet or attempt to forward it as is. Otherwise, non PPPOE internet access must be used if SNOM phones are to be used onsite.

Xten Lite & Eyebeam Softphone configuration

Wed, 18 Jun 2008 19:58:08 GMT
Recent version of Xten lite require some non-obvious settings to work properly when using Thinktel VoIP service. After launching Xten Lite, follow these steps exactly:

1.) Dial ***7469 and click Send.
2.) In the window that pops up, type 'rinstance' in filter. A number of values should pop up for this. For the entry labelled 'Proxy 0', change its value to 0 instead of 1.
3.) Close the window and save changes.
4.) Restart Xten lite.
5.) Configure the Xten for your Subscriber line. Set domain to vp.thinktel.ca, username to the 10 digit number and the appropriate password.
6.) Save changes and restart xten lite. Your number should be registered to the switch with active phone service.

Low volume on voicemail

Wed, 14 May 2008 05:06:14 GMT
If you have a SIP trunk and are using Asterisk for voicemail. we have found Asterisk records the volume quite low in its voicemail files.One approach is to use a post-voicemail command in the voicemail configuration, which can run Sox on the saved files and amplify them automatically.

A recommended level for amplification of these is -7dB from maximum.

How do I force uncompressed or compressed audio on my linksys ata adapter?

Wed, 23 Apr 2008 02:58:01 GMT

To force uncompressed audio (G711), tap *027110
To force compressed audio (G729), tap *02729

Can not receive calls

Wed, 23 Apr 2008 02:54:31 GMT

Problem:
Customer is unable to receive inbound calls.

Solution:
This is related to the IP Handset (IP Centrex), or the Gateway Device (IBL, PRI) ability to register to Thinktels network. There are a few possibilities:
1. Power Outages
2. Loss of Internet Connectivity
3. Router / Firewall security options are blocking VOIP traffic.
Note: Calls should still route to Auto Attendent and/or VM. Customer can also choose emergency routing options, i.e. route to cellular phones.

Call are Dropping but only sometimes

Wed, 23 Apr 2008 02:44:11 GMT

This may indicate that the phone is not reregistering on its set interval with the Thinktel switch (3600s) . The switch will assume that the device is no longer present and will drop any calls that may be in progress.
- Verify the latest firmware is being ran on the phone/adapter.
- If you are running your phone behind a NAT firewall that is SIP aware, packet mangling may be occuring.
- If you have other IP handsets or ATAs, check if they are also configured with the same phone number. This could create registration conflicts and sudden call drops while the other devices registers on the same number.

Calls drop 30 seconds into call

Wed, 23 Apr 2008 02:43:36 GMT
This may occur if you are using an ATA or IP Phone peered directly to Thinktel's switch.
What happens is our switch sends a SIP 'keep alive' every 30 seconds to make sure the call is still active. It does this be sending a SIP re-invite for the live call, in which we expect a 200 OK respsonse, that we then ACK back. If no response is receive, the call is terminated by the switch.

This could be caused by:

- A SIP aware firewall that is mangling packets.
- NAT issues on your router and does not support SIP properly

Asterisk & DTMF

Wed, 23 Apr 2008 02:43:19 GMT

Asterisk normally handles RFC2883 in a strange manner:

From the phone:
Audio Audio Audio DigitStart Duration Duration DigitEnd Audio Audio

It waits until it sees 'DigitEnd' before it does anything, then it sends (in a tight for loop):

DigitStart Start Start End End End (takes about 1ms normally)

So, this consumes the DTMF entirely (duration is lost), and it sends it in such a short period that devices that use the other definition of how long to make DTMF (you can either use Duration field, the Asterisk way, or the start playing with the start message, stop at the end way, which our switch does) end up generating the worlds shortest DTMF sequences.

Thus, the side effect of this patch, by slowing the packet rate down to around 80ms for the entire period, is it won't accept further DTMF/audio for those 80ms (Asterisk, that is, it will delay everything by that amount). So, really fast DTMF dialers can miss a keystroke.
For SIP peers running Asterisk 1.0.x - 1.2.x, a patch can be provided here

Calls are Dropping after a few seconds of being answered

Wed, 23 Apr 2008 02:43:00 GMT

This typically means there is no reply to the 200 OK from the Thinktel switch. The 200 OK indicates the call has been answered.
The phone could possibly be:
- Responding to the 200 OK but is routing to the wrong place.
- The phone is responing on the wrong port. ie: port 0
- The phone is responding back to the wrong IP address.
- A SIP aware firewall may be mangling packets or not forwarding SIP ACKs in NAT scenarios. We have seen this behavoir in Juniper firewalls.

Astra IP Phone service is up and down.

Wed, 23 Apr 2008 02:42:36 GMT

Astra phones do not support a proper NAT keep alive (as of writting this article) and can cause fadding in and out of inbound calling.
details of fix will be posted soon.

How do I request a number(s) to be ported to Thinktel?

Wed, 23 Apr 2008 02:41:29 GMT

Thinktel can provide LNP (local number portability) to centres in our on-net footprint. To request a port, please complete the following form:
http://support.thinktel.ca/ports

To view our current on-net and off-net foot print, refer to the following Wiki article.

What are the supported codecs?

Wed, 23 Apr 2008 02:40:59 GMT

- G.711 - Pulse code modulation (PCM) of voice frequencies on an 64 kbps channel. mu-Law companding (PCMU) and a-Law companding (PCMA)
- G.726-32 / G.721 - 32 kbit/s adaptive differential pulse code modulation (ADPCM)
- G.729 - Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP), including annex A and B.

* Do not use G.726 with Asterisk/Trixbox/CallWeaver