SIP forwarding problem

I have set up my extension 94086 to dial SIP/11109955030@67.215.241.250. However the system log reports an error 484 and I get an announcement. I can make a call to the above SIP URI using a soft phone registered to another pbxes.com extension.

Does the SIP URI feature still exist?

Thanks for the help.

RE: SIP forwarding problem

Well, calling the same SIP URI from my account, I get ring-back and a timeout since no-one answers the call.

Mar 4 16:26:21 VERBOSE[6235] logger.c: -- Called 11109955030@67.215.241.250
Mar 4 16:26:21 VERBOSE[13963] chan_sip.c: SIP response 100 to standard invite
Mar 4 16:26:21 VERBOSE[13963] chan_sip.c: SIP response 180 to standard invite
Mar 4 16:26:21 VERBOSE[6235] logger.c: -- SIP/67.215.241.250-5919 is ringing
Mar 4 16:26:56 VERBOSE[6235] logger.c: -- Nobody picked up in 35000 ms
Mar 4 16:26:56 VERBOSE[6235] chan_sip.c: Hangup call SIP/67.215.241.250-5919, SIP callid 37da837f782fef6938eb8eab66fbdc45@67.231.245.210
Mar 4 16:26:56 VERBOSE[6235] chan_sip.c: Hangup call SIP/Diafora-xxxx-736f, SIP callid 2183d81f-e428b750@xxx.xxx.xxx.xxx (Public IP of my phone)

RE: SIP forwarding problem

I can reach the SIP URI from a soft phone without any problems. However I don't seem to be able to FORWARD the call to the SIP URI. I created an extensions for the the purpose of forwarding and entered in SIP URI in the dial box.

Here is a link to the extension dialog screenshot: screenshot extension dialog

Here is the system.log for one of the failed attenpts.

Thanks, Bernd

Mar 4 04:51:10 VERBOSE[5263] chan_sip.c: SIP response 200 to standard invite
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found RTP audio format 0
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found RTP audio format 100
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found RTP audio format 101
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found RTP video format 34
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found RTP video format 103
Mar 4 04:51:10 VERBOSE[5263] logger.c: Peer audio RTP is at port 71.198.23.3:16414
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found description format PCMU
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found description format NSE
Mar 4 04:51:10 VERBOSE[5263] logger.c: Found description format telephone-event
Mar 4 04:51:10 VERBOSE[5263] logger.c: Capabilities: us - 0x18061e (gsm|ulaw|alaw|g726|speex|ilbc|h263|h263p), peer - audio=0x4 (ulaw)/video=0x180000 (h263|h263p), combined - 0x180004 (ulaw|h263|h263p)
Mar 4 04:51:10 VERBOSE[5263] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Mar 4 04:51:10 VERBOSE[5263] chan_sip.c: Oooh, we need to change our formats since our peer supports only 0x180004 (ulaw|h263|h263p) and not 0x40 (slin)
Mar 4 04:51:11 VERBOSE[18621] logger.c: We're at 76.191.104.53 port 37848
Mar 4 04:51:11 VERBOSE[18621] logger.c: Video is at 76.191.104.53 port 42510
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding codec 0x4 (ulaw) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding codec 0x8 (alaw) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding codec 0x10 (g726) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding codec 0x400 (ilbc) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding codec 0x2 (gsm) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding codec 0x200 (speex) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 4 04:51:11 VERBOSE[18621] logger.c: -- Called VoipMS/894086
Mar 4 04:51:11 VERBOSE[5263] chan_sip.c: SIP response 407 to standard invite
Mar 4 04:51:11 VERBOSE[5263] logger.c: We're at 76.191.104.53 port 37848
Mar 4 04:51:11 VERBOSE[5263] logger.c: Video is at 76.191.104.53 port 42510
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding codec 0x4 (ulaw) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding codec 0x8 (alaw) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding codec 0x10 (g726) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding codec 0x400 (ilbc) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding codec 0x2 (gsm) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding codec 0x200 (speex) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 4 04:51:11 VERBOSE[5263] logger.c: -- Got SIP response 484 "Address Incomplete"
Mar 4 04:51:11 VERBOSE[18621] chan_sip.c: Hangup call SIP/VoipMS-74bd, SIP callid 1a19022e39a9ffde0c71ea5d37473607@sip.us3.voip.ms
Mar 4 04:51:11 VERBOSE[18621] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
Mar 4 04:51:11 VERBOSE[18621] logger.c: -- Playing 'cannot-complete-as-dialed' (language 'en')
Mar 4 04:51:13 VERBOSE[18621] logger.c: -- Playing 'pls-try-call-later' (language 'en')

This post has been edited 5 time(s), it was last edited by ber on 05.03.2010 at 06:15.

RE: SIP forwarding problem

Bernd, I believe there is some misunderstanding between what you asked about in your first message, and the log you included in your last message.

Calling 894086 via your VoipMS trunk is entirely different from calling the 11109955030@67.215.241.250 SIP URI via the native OnNet trunk.

To recreate your setup, I provisioned one of my extensions with the SIP/11109955030@67.215.241.250 in the "dial" field identically to your screenshot, I was able to call it from another SIP extension and pasted the relevant part of my System Log.

How does using VoipMS relate to this inquiry?

RE: SIP forwarding problem

Thanks for looking into this!

I just double checked my call logs. My call log shows yesterdays call from "SPA962p4@Run" around 01:26 PM PST. If you called today,
your call did NOT get through.

SIP SIP/11109955030@67.215.241.250 is at voip.ms, the equivalent address is SIP/11109955030@sip.us3.voip.ms.
I tried to use my VoipMs outgoing trunk because I was hoping it would go through since trunk and target are with the same provider. I'm open to any other suitable setup which could make this work.

Thanks, Bernd

RE: SIP forwarding problem

Well, I dialed another two calls today to the URI with the FQDN you mentioned, via an extension setup on my account. I got the same result as before. Ring No Answer.

Mar 9 06:59:56 VERBOSE[31109] logger.c: -- Called 11109955030@sip.us3.voip.ms
Mar 9 06:59:56 VERBOSE[23968] chan_sip.c: SIP response 100 to standard invite
Mar 9 06:59:56 VERBOSE[23968] chan_sip.c: SIP response 180 to standard invite
Mar 9 06:59:56 VERBOSE[31109] logger.c: -- SIP/sip.us3.voip.ms-cb1a is ringing
Mar 9 07:00:31 VERBOSE[31109] logger.c: -- Nobody picked up in 35000 ms
Mar 9 07:00:31 VERBOSE[31109] chan_sip.c: Hangup call SIP/sip.us3.voip.ms-cb1a, SIP callid 2e07efbf0abf710f4f0cd0e47c5d9075@67.231.245.210
Mar 9 07:00:31 VERBOSE[31109] chan_sip.c: Hangup call SIP/Diafora-xxxx-2176, SIP callid 728e7511-bd6ec130@xxx.xxx.xxx.xxx (Public IP of my phone)

Mar 9 08:42:09 VERBOSE[27715] logger.c: -- Called 11109955030@sip.us3.voip.ms
Mar 9 08:42:09 VERBOSE[23968] chan_sip.c: SIP response 100 to standard invite
Mar 9 08:42:09 VERBOSE[23968] chan_sip.c: SIP response 180 to standard invite
Mar 9 08:42:09 VERBOSE[27715] logger.c: -- SIP/sip.us3.voip.ms-8773 is ringing
Mar 9 08:42:44 VERBOSE[27715] logger.c: -- Nobody picked up in 35000 ms
Mar 9 08:42:44 VERBOSE[27715] chan_sip.c: Hangup call SIP/sip.us3.voip.ms-8773, SIP callid 014f11eb2c61cc56077fcf526c04827b@67.231.245.210
Mar 9 08:42:44 VERBOSE[27715] chan_sip.c: Hangup call SIP/Diafora-9624-e130, SIP callid a347db6-d462a451@xxx.xxx.xxx.xxx (Public IP of my phone)

As far as I can tell, both of my calls got through, but there was no answer. That is quite different from claiming the calls didn't get through, for whatever reason.

So the questions to resolve this are: What is on the other side of the SIP URI? Who or what is supposed to answer the inbound calls?

RE: SIP forwarding problem

The SIP address in question is a virtual DID to my voip.ms account. The DID is routed to my account which is registered to a PAP2T in my home where it rings my phone.

While I can not guarantee the accuracy of voip.ms' call detail records, so far they have been very accurate. Your calls this morning did go through. The times in the CDR for the two calls are

03:59:56 PST and 05:42:09 PST. I could not answer the call because I was sleeping.

Encouraged by your reply I set up a new extension (deleted all previous attempts beforehand) but unfortunately I only got a busy signal. I made several attempts. Inbetween two attempts to reach the extension I managed to get through with my softphone calling the virtual DID directly.

Here are the last few lines of the system.log:
Mar 9 16:13:57 VERBOSE[16422] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Mar 9 16:13:57 VERBOSE[16422] logger.c: -- Called 11109955030@sip.us3.voip.ms
Mar 9 16:13:57 VERBOSE[27156] chan_sip.c: SIP response 407 to standard invite
Mar 9 16:13:57 NOTICE[27156] chan_sip.c: Failed to authenticate on INVITE to '"4087304823" ;tag=as3861de32'
Mar 9 16:13:57 VERBOSE[16422] logger.c: -- SIP/sip.us3.voip.ms-8285 is circuit-busy
Mar 9 16:13:57 VERBOSE[16422] chan_sip.c: Hangup call SIP/sip.us3.voip.ms-8285, SIP callid 5b67c2b04b9bf44d2197c030226efac5@76.191.104.53
Mar 9 16:13:57 VERBOSE[16422] logger.c: == Everyone is busy/congested at this time (1:0/1/0)
Mar 9 16:13:57 VERBOSE[16422] chan_sip.c: Hangup call SIP/1503996-346f, SIP callid 301826028-3477168837-179212@gsbc04-lsan.mdsg-pacwest.com

Thanks, Bernd