Welcome to the Cisco Support Community Ask the Expert conversation. This is an opportunity to learn about resolving usage, configuration, and troubleshooting of SPA phones and ATAs.Patrick Born is a Technical Marketing Engineer for the Small Business Technology Group (SBTG). His responsibilities include SPA3xx, SPA5xx, and SPA9xx IP Phones, WIP 3xx IP Phones, and SPAxxx ATAs including the SPA2102, SPA3102, PAP2T devices.
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Patrick might not be able to answer each question due to the volume expected during this event. Remember that you can continue the conversation on the Small Business sub-community discussion forum shortly after the event. This event lasts through November 18, 2011. Visit this forum often to view responses to your questions and the questions of other community members.
[I previously responded but had mangled the message. To eliminate confusion, I've deleted my previous post.]
I can confirm that IPv6 is being investigated for the SPA30x, SPA50x, and SPA525G2 IP phone roadmap.
You might be able to help me with my issue. I have a small office with 10 phones 5 at 2 sites. The current configuration consists of a RV016 connected to a RV082 VPN routers. We have a Asterisk Box and SPA942 phones. The current network works nicely.
We decided to upgrade to a 2911 with a 24 port POE switch in it from the RV082 since we are going to be routing multiple IP addresses, Soon as we replace and setup the 2911 all RTP traffic drops from the network. On the local site the PBX is hooked up directly to the Switch, that has a base address in the 192.168.1.x range, the remote site is through a VPN and is the 192.168.0.x range.
I am not a noobie when configuring the 2911, but I must be missing something. Truthfully, I do not care about the call manager or any of the advanced features. All i need is to route 3 outside addresses in the local office and have the remote site connect with the PBX. Data communications work fine, but all voice traffic from the remote site is dropped or ignored.
I have some questions about the 2911 and I pray that you can help. I am at the end of my rope here. I have spoken with 25 different reps, and only 1 seems to really care about the issue.
I have tried the following:
Deny the 192.168.0.x subnet from the nat
Creating access lists for UDP ports 5060 and 10000-20000
Everything in Cisco Document 5162
No ip nat inside service voip 5060 (Disable ALG)
Kicking the router!
1) Does the 2911 default to not allowing RTP traffic to pass if you do not have a firewall set?
2) Do I have to use the CME with a asterisk box?
3) How can I setup this router to be in dumb mode turn everything off and not anaylize traffic?
4) What would you recommend doing. I am so burnt out on this I can't begin to tell you....Please please help.
We use SPA-942 phones with the G711ulaw codec. Might you recommend using another, if possible. Is there any special settings we need to have setup for the traffic. Please let me know thank you.
I'm no 2911 or IOS expert so can't offer specific help with this.
One new variable new to the mix could be that the 2911 and / or switch may be supporting the Cisco Discovery Protocol (CDP).
The easiest way to test if this is a factor is to disable CDP on the SPA942 phones either via the phone's web-UI or through their configuration file.
There is no need to supplement Asterisk with any other call control.
There is no real need to move from G.711 ulaw to another CODEC.
Keep in mind that when a call is established, the SIP signaling path is usually not the same path that the RTP traffic will take. This makes me think that there may be stateful packet inspection being performed by a firewall which results in RTP not being expected and consequently dropped, resulting in no RTP / on-way audio.
In summary, I believe that you are facing two issues, CDP and a firewall.
Best of luck. Please share with the community and me, your progress,
The SPA504G has 4 line buttons in the form of multi-colored LED buttons.
Each line allows a unique SIP registration or can share a registration with another line.
The phone is capable as of firmware release 7.4.8 of supporting 10 call appearances across any of the line keys in any way you'd like.
For example, you may choose to only register line key 1 and you could be speaking to person 10 while persons 1 through 9 are on hold.
Great question, thanks for asking.
Patrick Born | Technical Marketing Engineer (TME) – SPA SIP Endpoints
Use this reference document to locate SPA phone resources
i have doubt i got a cisci id but its a 8 digit id i heard cisco id is total 12 digit is it true??
and i want to know how to check and verifiy about certifiction status???help me out
I don't work with Cisco IDs or certifications.
What certification are you wanting? Perhaps I can direct you to someone who can help better than I can.
I am having two issues at a customer site I am hoping you could give me some insight into. They have a UC320 system with one SPA508 phone with side car and the other phones are SPA303s. The issues are:
1) The customer is complaining that the phones hang up or "freeze" and become unresponsive to input at random periods. It does not happen with all phones at the same time. Its one phone at a time and it is random. I have seen it where if you press a key you get a high pitch tone and the phone is unresponsive. You have to reboot the phone for it to work. I noticed this behaviour after I upgraded the UC320 firmware to version 2.1.1. This also occurs with the SPA508 phone as well.
2) The other issue is the the time that the system takes to actually outpulse the digits entered on the phone's key pad and the dial soft key is pressed. Is this something that can be reduced??
These issues have the customer at the point of wanting to return the system. Any ideas would be greatly appreciated.
Thanks for you feedback.
1. I request to kindly take the time to open a SBSC case on this issue. Phone freezes are rare, however when they happen we want to know everything about them. I request you to kindly send me the exact circumstances under which the phone froze to email@example.com.
2. The phone typicaly takes 3-4 secs to send digits to the Telco. Does your Telco play a stuter tone? This may delay the Post Dial Delay.
Thank you for the assistance I will definitely act on your suggestions. With regards to issue 2, no the Telco does not play a stutter tone. It takes closer to 10 seconds to actually hear ring back from the Telco instead of 3-4 like you mentioned.