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jjp
Registration Date: 01.01.1970
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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.
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07.06.2008 23:23 |
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jjp
Registration Date: 01.01.1970
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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.
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11.06.2008 00:33 |
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Dia
Premium Account
Registration Date: 03.03.2006
Posts: 1443
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Well it's not really broken, is it? |
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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.
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12.06.2008 02:23 |
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