It is a common problem that people starting out with Asterisk PBX find it difficult to diagnose where problems arise. Here is a quick blog that may help someone in that situation.
To debug asterisk you need to get hold of the asterisk messages and any other log info you can. This isn't always so simple if you have a busy system as so many records are created.
If you have a lot of calls its important to be very quick when you place a test call and capture output so you won't have multiple calls in the same capture.
Log into your asterisk server via ssh.
Usually I then type :
asterisk -rx "sip set debug on"
asterisk -rcvvvvvvvvvvvvvvvvvvvvv> /usr/local/admin/htdocs/debug.txt
(note 1: you can use asterisk -rx "sip set debug on IP" if you know a particular IP, or use PEER. Type help from the command line for options)
(note 2: replace /usr/local/admin/htdocs/debug.txt with some path on your server. If the path is to htdocs then you can easily download the capture using a browser)
And after that I call the number that I want to test and then I hangup when the call goes through. Then I quickly stop the asterisk capture with CTRL+C.
After that I verify the Asterisk capture. I just download it and view in a text editor.
In this situation I only verified the Asterisk log but in another situation I might have to check the SIP packets. To do that I use tcdump.
If tcpdump is not installed you can run 'yum install tcpdump' on a linux system to install.
I then run something like
'tcpdump -p udp -s 4096 -w /usr/local/admin/htdocs/capture.pcap'
to capture all traffic.
or, 'tcpdump -w /usr/local/admin/htdocs/capture.pcap -s 0 host sip.avoipcarrier.net' to get a dump for just one host.
After I place a call from an extension to an specific number I stop the capture with CTRL+C and I download the capture to my computer.
Then using a tool like Wireshark (you can find that on the internet since its a free tool) I verify the call flows and view the SIP messages.
I guess this is a basic howto on how to debug basic things. if you want to learn more check some articles regarding basic call flows and how SIP handles basic transactions.
Also check some tutorials about Wireshark (source/destination port/address, how to debug rtp, sip messages, filters, jitter...).
I hope this helps.
Monday, June 28, 2010
Thursday, June 24, 2010
Using 3CXphone
3CXPhone is a free VoIP phone that allows you to use your PC or laptop as a phone.
You can connect 3CXPhone to a VOIP provider such as mydivert.com to make calls to any VoIP, mobile or landline number. By adding an incoming virtual number to your mydivert.com account you can be instantly contactable on a local number from over 65 countries worlwide.
To download 3cxphone simply visit the 3cx website. The download is 3MB and installation only takes a couple of minutes.
The 3CXphone compares very favourably to other softphones. Time to establish SIP registartion is fast and dialed calls are established almost immediatly. The phone interface is attractive and user friendly. All the usual features are provided such as voicemail alerts, address book, call records etc.
To configure 3CXphone with mydivert.com you should create a new SIP profile and add your account details as below:
The advanced settings can then be set. We added *95 as the voicemail access number, and checked that the SIP transport method is set to UDP (important!). As we are running the sofphone nehind a NAT router we set a short registration period (2 - 5 mins) to help with keep-alives and maintain the phoen presence.
As we want the phone to send DTMF tones that we will use when logging into out voicemail system we unchecked the INBAND and SIPINFO DTMF signaling. These methods are not supported.
That's all. The softphone works very well and we would rate it with 5 stars.
3CX also offer complete pbx systems for windows which work well with our sip trunking service.
You can connect 3CXPhone to a VOIP provider such as mydivert.com to make calls to any VoIP, mobile or landline number. By adding an incoming virtual number to your mydivert.com account you can be instantly contactable on a local number from over 65 countries worlwide.
To download 3cxphone simply visit the 3cx website. The download is 3MB and installation only takes a couple of minutes.
The 3CXphone compares very favourably to other softphones. Time to establish SIP registartion is fast and dialed calls are established almost immediatly. The phone interface is attractive and user friendly. All the usual features are provided such as voicemail alerts, address book, call records etc.
To configure 3CXphone with mydivert.com you should create a new SIP profile and add your account details as below:
The advanced settings can then be set. We added *95 as the voicemail access number, and checked that the SIP transport method is set to UDP (important!). As we are running the sofphone nehind a NAT router we set a short registration period (2 - 5 mins) to help with keep-alives and maintain the phoen presence.
As we want the phone to send DTMF tones that we will use when logging into out voicemail system we unchecked the INBAND and SIPINFO DTMF signaling. These methods are not supported.
That's all. The softphone works very well and we would rate it with 5 stars.
3CX also offer complete pbx systems for windows which work well with our sip trunking service.
SIP Trunking for PBX
We have introduced a new feature to the user account area allowing the user to enable/disable the SIP Trunking Service for PBX users.
This feature will provide additional flexability to our system, allowing the use of SIP accounts with standard VoIP endpoint equipment, extension telephone devices and softphones etc, as well as PBX servers such as Asterisk and FreePBX. Previously the SIP Trunking Service had to be enabled by our support engineers on request.
What is a SIP Trunking Service?
Sounds very technical, and expensive.... like a lot of the termanology that surrounds VoIP. In truth, it is very simple, and doesn't cost a thing.
SIP Trunking for PBX just sends call invites in a different form so that a PBX knows on which DID number a call arrives and can route it to the correct endpoint.
Invite for PBX : DIDNUMBER@IP:port
Invite for extension device : SIPUSERNUMBER@IP:port
It is important that the correct invite is sent for the correct equipment else incoming calls are not received.
By adding a feature to 'toggle' between invite forms we enable both possabilities.
SIP Trunks can work with any SIP-ready PBX - contact us if you have any questions.
This feature will provide additional flexability to our system, allowing the use of SIP accounts with standard VoIP endpoint equipment, extension telephone devices and softphones etc, as well as PBX servers such as Asterisk and FreePBX. Previously the SIP Trunking Service had to be enabled by our support engineers on request.
What is a SIP Trunking Service?
Sounds very technical, and expensive.... like a lot of the termanology that surrounds VoIP. In truth, it is very simple, and doesn't cost a thing.
SIP Trunking for PBX just sends call invites in a different form so that a PBX knows on which DID number a call arrives and can route it to the correct endpoint.
Invite for PBX : DIDNUMBER@IP:port
Invite for extension device : SIPUSERNUMBER@IP:port
It is important that the correct invite is sent for the correct equipment else incoming calls are not received.
By adding a feature to 'toggle' between invite forms we enable both possabilities.
SIP Trunks can work with any SIP-ready PBX - contact us if you have any questions.
Wednesday, June 23, 2010
virtuellt nummer Sverige
virtuellt nummer Sverige - alla riktnummer tillgangliga .... which we believe translates as "virtual number sweden - all area codes available" !!
However it translates, we are still very pleased to introduce a complete numbering range of local area codes from our Swedish VoIP partners.
These DID are excellent quality and enabled with 5 channels for incoming calls.
Two National area codes are also available, 46-10 and 46-77.
Each DID number is billed on a fixed monthly fee basis with no per-minute charges. Incoming SIP calls are free. The call forwarding destination of the virtual number, or SIP URI mapping, can be updated at any time via the user account area at mydivert.com
- enjoy
However it translates, we are still very pleased to introduce a complete numbering range of local area codes from our Swedish VoIP partners.
These DID are excellent quality and enabled with 5 channels for incoming calls.
Two National area codes are also available, 46-10 and 46-77.
Each DID number is billed on a fixed monthly fee basis with no per-minute charges. Incoming SIP calls are free. The call forwarding destination of the virtual number, or SIP URI mapping, can be updated at any time via the user account area at mydivert.com
- enjoy
Australian Virtual Numbers
We are pleased to introduce additional local area codes for our Australian virtual number coverage.
The additional local area codes are;
Australian DID area - Geelong: +61 352
Australian DID area - Gosford: +61 243
Australian DID area - Launceston: +61 363
Australian DID area - Thuringova: +61 747
Australian DID area - Wollongong: +61 242
These Australian DID virtual numbers are now available for immediate purchase at mydivert.com.
The additional local area codes are;
Australian DID area - Geelong: +61 352
Australian DID area - Gosford: +61 243
Australian DID area - Launceston: +61 363
Australian DID area - Thuringova: +61 747
Australian DID area - Wollongong: +61 242
These Australian DID virtual numbers are now available for immediate purchase at mydivert.com.
Tuesday, June 22, 2010
Nederland Virtual Numbers
A complete Geographical Virtual Number range is now available at mydivert.com for the Netherlands (Nederland).
All Dutch Virtual Numbers are enabled with up to 10 channels for incoming sip calls and cost only €5.50 per month.
Now you can have telephone numbers from virtualy anywhere in the Netherlands that can be directed to (ring-to) any phone or sip uri in the world.
Free Unlimited Incoming SIP Calls
No Setup Fees - Cancel Anytime
A local address corresponding to relevant geographic zone is required for Nederland DID numbers.
All Dutch Virtual Numbers are enabled with up to 10 channels for incoming sip calls and cost only €5.50 per month.
Now you can have telephone numbers from virtualy anywhere in the Netherlands that can be directed to (ring-to) any phone or sip uri in the world.
Free Unlimited Incoming SIP Calls
No Setup Fees - Cancel Anytime
A local address corresponding to relevant geographic zone is required for Nederland DID numbers.
Thursday, June 17, 2010
Bria Softphone for Iphone
CounterPath, the makers of the popular X-Lite and Bria softphone, just went mobile with their SIP application for the Apple Iphone.
The Bria iPhone Edition enables enterprise-class security with call signaling and audio encryption via TLS and SRTP. This makes the softphone a first of its kind.
Bria iPhone Edition also integrates seamlessly with the Desktop softphone application.
Bria iPhone Edition Features include:
Full SIP compliance, enabling use with any SIP-compliant server.
Call quality that's superior to circuit-switched wireline and wireless voice, and based on the G.711, G.729 and GSM codecs.
An intuitive user interface that provides one-touch access to voicemail, call history, speakerphone and other frequently used telephony features such as 4-digit dialing.
Support for DTMF, which lets you enter numbers to access an auto attendant.
Multi-call management options, including the ability to switch between, merge and split calls, all in ways that are already familiar to iPhone users so you can start using these features immediately.
Take the exceptional voice quality, security and usability of your Bria desktop to your iPhone, iPad or iPod touch and make free and low-cost calls over your 3G or Wi-Fi network to anyone, anywhere.
Take advantage of the limited-time introductory pricing of only $3.99 USD!
View Bria Iphone Edition at www.counterpath.com
The Bria iPhone Edition enables enterprise-class security with call signaling and audio encryption via TLS and SRTP. This makes the softphone a first of its kind.
Bria iPhone Edition also integrates seamlessly with the Desktop softphone application.
Bria iPhone Edition Features include:
Full SIP compliance, enabling use with any SIP-compliant server.
Call quality that's superior to circuit-switched wireline and wireless voice, and based on the G.711, G.729 and GSM codecs.
An intuitive user interface that provides one-touch access to voicemail, call history, speakerphone and other frequently used telephony features such as 4-digit dialing.
Support for DTMF, which lets you enter numbers to access an auto attendant.
Multi-call management options, including the ability to switch between, merge and split calls, all in ways that are already familiar to iPhone users so you can start using these features immediately.
Take the exceptional voice quality, security and usability of your Bria desktop to your iPhone, iPad or iPod touch and make free and low-cost calls over your 3G or Wi-Fi network to anyone, anywhere.
Take advantage of the limited-time introductory pricing of only $3.99 USD!
View Bria Iphone Edition at www.counterpath.com
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