Several annoyances

Hi,

I am experiencing two problems that i can't seem to be able to resolve. I have tried multiple outbound providers.

First when I place an outbound call I will first hear a tone (sometimes for as much as 40 seconds!) before I get connected. The tone I think is produced by my phone to signal the connection is not being set up. Then sometimes when the tone ends and the call is set up, I start out in the middle of the voicemail greeting of the person I'm calling.

Second, I get complains from people that I leave them 1 minute long SILENT voicemail messages. Then after some research I found out that I did not even leave those people a VM message and hung up before getting to the tone. When testing this by calling one of my own DID's and monitoring the STATUS screen I noticed that when I hang up the timer on the status screen keeps running, sometimes for as much as another 90 seconds!!!! Also when I place a call to a regular pstn line the phone keeps ringing way after I hang up the phone.

I have tested and confirmed this behavior on several phones, several extension, different termination providers and from different client ISP's.

Thanks for your response

This post has been edited 1 time(s), it was last edited by jjp on 08.06.2008 at 01:52.

These are indeed annoying issues.

Regarding the first issue, what kind of SIP UAs do you get this behavior with? According to my experience, this sounds like a "feature" of Siemens SIP phones.

Regarding the second issue, of which countries and cities PSTN numbers' are you dialing, when you are experiencing this issue? Also, on which server is your account hosted?

Thanks for your response!

You're right a Siemens phone is producing the tone. I'd assume it does that because the server is not setting up the connection promptly, right? That is, when I replicate the behavior on a Softphone, I do not hear the tone, but rather I hear nothing until the connection is setup. Then again, sometimes the connection is setup straight away but the person I call cannot hear me until like 10 seconds after they've picked up. Like I said, sometimes when I get someone's voicemail, I do not hear anything until halfway through the greeting.

I experience these issues when I dial any number. I do not have the impression it is country related. Mostly however, I use voip to call people in the US. The account is hosted on the www5 server. I changed it to www1 to see if that changed anything, I can now report that it doesn't. I don't recall having these problems when I was on the Frankfurt server, but you guys closed that one on customers using 2gb+ a month. In retrospect, I do kinda think that was a bad move, especially without any advance notice or anything. I'm not saying this particular problem is related to that though!

Thanks!

This post has been edited 1 time(s), it was last edited by jjp on 08.06.2008 at 01:54.

What I have noticed with Siemens SIP phones, is they sometimes are not sending the call to the SIP Proxy (wwwX.pbxes.com) at all, even if they seem to be registered to it. Sometimes the call will get through on the 2nd or 3rd attempt, other times it will not.

What I have discovered is that, if I access the web interface of the phone, not the base, the call might go through. Other times I have to de-register and re-register the particular extension by clicking on the appropriate checkbox.

Regarding the rest of the issues, you will need to setup some SIP traces in order to pinpoint the issue from there. Do you have any other SIP UAs available except the Siemens?

Hi!

Thanks again! I currently do not have any other UA's here at home. I tried replicating the issue with a professional SIP phone from work but it worked perfectly. So can you recommend me a good UA to replace the Siemens that offers reasonable value? Or do you have suggestions to make the Siemens phone work properly using PBXes? It might be something worth looking in to? I can hardly imagine Siemens shipped out millions of broken SIP phones, right? Thanks!

I will add the log of call that i just placed to GOOG 411 (took out some id's that might contain passwords or personal data (not sure they do, just wanted to be sure)). The behavior was exactly as described. That is, I did not get connected, but rather the Siemens phone produced the weird tone. Do you see any abnormalities?

Jun 11 00:34:25 VERBOSE[13221] logger.c: Found RTP audio format 0
Jun 11 00:34:25 VERBOSE[13221] logger.c: Found RTP audio format 101
Jun 11 00:34:25 VERBOSE[13221] logger.c: Peer audio RTP is at port [homeIP]:26996
Jun 11 00:34:25 VERBOSE[13221] logger.c: Peer video RTP is at port [homeIP]:65535
Jun 11 00:34:25 VERBOSE[13221] logger.c: Found description format PCMU
Jun 11 00:34:25 VERBOSE[13221] logger.c: Found description format telephone-event
Jun 11 00:34:25 VERBOSE[13221] logger.c: Capabilities: us - 0x18041e (gsm|ulaw|alaw|g726|ilbc|h263|h263p), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Jun 11 00:34:25 VERBOSE[13221] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Jun 11 00:34:25 VERBOSE[2831] logger.c: We're at 91.121.136.13 port 42170
Jun 11 00:34:25 VERBOSE[2831] logger.c: Video is at 91.121.136.13 port 41834
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x8 (alaw) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x10 (g726) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x400 (ilbc) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding codec 0x2 (gsm) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 11 00:34:25 VERBOSE[2831] logger.c: -- Called JP_VoIP.ms/8004664411
Jun 11 00:34:25 VERBOSE[13221] chan_sip.c: SIP response 407 to standard invite
Jun 11 00:34:25 VERBOSE[13221] logger.c: We're at 91.121.136.13 port 42170
Jun 11 00:34:25 VERBOSE[13221] logger.c: Video is at 91.121.136.13 port 41834
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x4 (ulaw) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x8 (alaw) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x10 (g726) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x400 (ilbc) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding codec 0x2 (gsm) to SDP
Jun 11 00:34:25 VERBOSE[13221] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 11 00:34:25 VERBOSE[13221] chan_sip.c: SIP response 100 to standard invite
Jun 11 00:34:39 VERBOSE[2831] chan_sip.c: Hangup call SIP/JP_VoIP.ms-325b, SIP callid [I D @ SERVER . COM]
Jun 11 00:34:39 VERBOSE[2831] chan_sip.c: Hangup call SIP/[A C C O U N T]-0001-f4c0, SIP callid [I D]@[HOME_IP]

This post has been edited 4 time(s), it was last edited by jjp on 11.06.2008 at 00:50.

Well it's not really broken, is it?

It's not easy at all to recommend another DECT SIP UA, since I am only aware of just 2 phones being available in the market. So, if you are looking for a DECT SIP UA it's either one of the Siemens phones or the Snom M3.

Although I have not worked with the Snom M3 directly, I can confirm the desktop Snom 360 phones, have a few issues on their own with PBXes, chiefly loosing their registration and not accepting incoming calls intermittently.

It should also be noted, that although I have only been working with a Siemens CE460 IP R phone for the past month, I have been able to identify a few issues with the Siemens SIP stack. One of them is the phone stops responding to both incoming and outgoing call attempts.

Now regarding you comment, "I can hardly imagine Siemens shipped out millions of broken SIP phones, right?", you should consider a few aspects related to their philosophy when building products in general:

• "The phone is not broken, since by definition phone is the PSTN part of it, which works without any issues in the country it was legally sold. Consider the SIP part of it a bonus, when it works". This can be illustrated, by their conscious decision to produce only 2 models (CE460 IP R and CE450 IP R) with pure SIP functionality, and to legally sell them exclusively in the Nordic region. These phones are not even sold in Germany, even if they lack any PSTN interface, instead of being available for sale in every country of the world with an available Internet connection. But they are not sold worldwide, just the opposite, and I have the Gigaset support email messages to prove it.

• "The phone is not broken, since it can be fixed via a firmware update". The SIP functionality is pretty recent on Siemens DECT phones, and their SIP stack is update-able via downloadable firmware updates. Germans in general (no racial remark intended) are known to keep improving their product offerings for a long time, while they keep perfecting every aspect of them. The Porsche 911 was officially introduced in 1964 and is being perfected ever since.

• "The phone is not broken, it's your service provider which is not compatible with our phones". In theory, any SIP UA should be compatible with any SIP Proxy, right? As you have undoubtedly found out that is not always the case. But Siemens has a point here, since their website lists all the providers their Gigaset phones have been certified to work with. PBXes is not among them. If we want to change that, PBXes needs to go through the Siemens Gigaset certification process, whatever that is. It shouldn't be hard, since we have our Gigaset phones working with PBXes already.

Not to contradict (yours was an opinion I think) but to document: I have a C450 IP bought in Bahrain and it worked out of the box. I do not remember when I bought it but it must have been early this year.
The Firmware version is
010700000000 / 041.00
and the EEPROM version is
87

I have bought another and configured/delivered it to work out of Egypt - works fine..

So
1. Siemens is selling them worldwide
2. They actually work.

Zitat:

Originally posted by jjpieper Hi! ..... I will add the log of call that i just placed to GOOG 411 (took out some id's that might contain passwords or personal data (not sure they do, just wanted to be sure)). The behavior was exactly as described. That is, I did not get connected, but rather the Siemens phone produced the weird tone. Do you see any abnormalities? ...... SIP callid [I D]@[HOME_IP]

What is GOOG 411 ??

Some facts

• Siemens is selling their DECT handsets in specific countries, as PSTN phones and they work perfectly as such.
  
• Their VoIP functionality is a bonus if and when it works. The only non PSTN handsets available are the CE460 IP R and CE450 IP R, which are only available for sale in the Nordic countries.
  
• Not every DECT model is available for sale in every country. That is a fact and I have got the email messages from Gigaset's support to prove it, regarding the above mentioned models.
  

Thank you for the explanation. Mine is a C450 IP and it is both a PSTN and an IP phone. I must admit, it is not the best phone I have but it does serve a purpose. I was quite glad when it worked with pbxes on the first try. Am also glad that yours work too.

I still don't understand why the same software (firmware) does not work the same way but I've seen that on the Sipuras before. I cannot dedicate any time to trying to perfect the way the Siemens works because they do not give enough of an interface to manage the device. It is unfortunate because the concept is great but I would never try to standardize on them Siemens devices (SIP performance cannot be made optimal).