 Thread: RE: Dial rule change |
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Hi,
my SIP provider changed their dialing format. Now I have to send numbers in the international gsm form, like this plus sign, country code, number.
I thiught it would be as easy as changing the dial rules on the trunk to plus sign, country code, plus sign, dot.
I did just that, but the change does not seem to get committed. I tried changing the language on the given trunk so that I see if that changes the pls-try-call-later track but no, it is still in English. I did click APPLY.
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| Thread: RE: plus sign |
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I need a dialing rule to replace +32456789012 with 0456789012. How do I do that? TIA.
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| Thread: RE: Belgacom BBox |
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Diafora,
I do not think you read my posts very carefully.
My bbox connects, it rings on incoming calls and I can take these calls. I can tell you that there are a lot of SIP packets involved in this process without ethereal. My problem is that DTMF signals don't seem to go through for some reason. I don't suppose I will find a solution for this but I made a point of reporting the problem on this forum so that YOUR other users in the future might find it helpful.
The BBOX is otherwise a provider branded user premise adsl equipment and as such it is not intended for third party VoIP provider services. Reseting, reconfiguring or replacing is not really a viable option.
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| Thread: RE: Belgacom BBox |
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I thought it was, but now I tried to reproduce it and I can't: Nothing gets logged. I turned SIP trace on, still nothing -- obviously the above log is not for the dial attempt. On the status page (on the flash GUI) the call attempt does NOT get displayed either. Looks like the busy tone I get immediately after dialing is generated on the Bbox itself. Any suggestions?
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| Thread: RE: Belgacom BBox |
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REGISTRATION PROBLEM SOLVED.
I had to change the setting of VPI/VCI on the ATM Interface of MAC Encapsulated Routing (under ATM PVC) from 0/40 to 1/40.
UNFORTUNATELY,
There is another problem, though: Every time I dial out, I get a busy tone. Even if I just dial another extension. I thought it might be a problem with DTMF, I tried fiddling with the telephone settings: Disabled 'Support Out of Band DTMF', and tried different codecs (G729 and G.723.1) to no avail. I can call from 101 to 111 no probs, but calls from 111 to 101 (or to any number, really) get a busy response. Here is the log:
Oct 28 11:49:54 VERBOSE[106684] logger.c: -- SIP/luxapo-111-01c5 is ringing
Oct 28 11:49:54 VERBOSE[106669] logger.c: -- Local/111@from-internal-cont/n-c988,1 is ringing
Oct 28 11:50:20 VERBOSE[62714] chan_sip.c: SIP response 200 to standard invite
Oct 28 11:50:20 VERBOSE[62714] logger.c: Found RTP audio format 8
Oct 28 11:50:20 VERBOSE[62714] logger.c: Peer audio RTP is at port 87.64.23.165:5002
Oct 28 11:50:20 VERBOSE[62714] logger.c: Peer video RTP is at port 87.64.23.165:65535
Oct 28 11:50:20 VERBOSE[62714] logger.c: Found description format PCMA
Oct 28 11:50:20 VERBOSE[62714] logger.c: Capabilities: us - 0x18061e (gsm|ulaw|alaw|g726|speex|ilbc|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Oct 28 11:50:20 VERBOSE[62714] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Oct 28 11:50:20 VERBOSE[62714] chan_sip.c: Oooh, we need to change our formats since our peer supports only 0x8 (alaw) and not 0x4 (ulaw)
Oct 28 11:50:20 VERBOSE[106684] logger.c: -- SIP/luxapo-111-01c5 answered Local/111@from-internal-cont/n-c988,2
Oct 28 11:50:20 VERBOSE[106669] logger.c: -- Local/111@from-internal-cont/n-c988,1 stopped sounds
Oct 28 11:50:20 VERBOSE[106669] logger.c: -- Local/111@from-internal-cont/n-c988,1 answered SIP/luxapo-101-b5b9
Oct 28 11:50:20 VERBOSE[106669] logger.c: We're at 88.198.69.250 port 39786
Oct 28 11:50:20 VERBOSE[106669] logger.c: Video is at 88.198.69.250 port 41336
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding codec 0x200 (speex) to SDP
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding codec 0x2 (gsm) to SDP
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding codec 0x100000 (h263p) to SDP
Oct 28 11:50:20 VERBOSE[106669] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Oct 28 11:50:22 VERBOSE[106684] chan_sip.c: Hangup call SIP/luxapo-111-01c5, SIP callid 3e136efd2d1bc88133e8bfae25572f89@88.198.69.250
Oct 28 11:50:22 VERBOSE[106669] chan_sip.c: Hangup call SIP/luxapo-101-b5b9, SIP callid 730981728392@172.24.80.168
Oct 28 11:50:26 VERBOSE[62714] logger.c: -- Registered SIP 'luxapo-101' expires 3600
Please advise.
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| Thread: RE: Belgacom BBox |
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I am trying to configure a Voip enabled BBOX router. It has the following SIP configuration page:
SIP Setting
Configure the following SIP-related parameters. And press Save button.
SIP Parameters
SIP Stack Mode NGN Stack IMS Stack
SIP UserAgent
UserAgent Domain:
UserAgent Port:
Proxy Setting
Proxy Domain:
Proxy Port:
Registrar Setting
Registrar Domain:
Registrar Port:
Outbound Proxy
Outband Proxy Domain:
Outband Proxy Port:
Re-Registration Time Interval
Support PRACK (RFC3262) Enable
Support SIP Session Timer (RFC 402 Enable
'Session Expires' timer:
'Minimum Session Expires' timer:
It also has a "Phone Number Setting" Page which is a little more straight forward:
Phone Number
Display Name
Realm
Username
Password
I would like to ask for some help with these settings, because registration fails. I tried both pbxes.org and 188.40.65.148 for User Agent, Proxy and Registrar settings with port 5060 everywhere.
On the status page I see this :
SIP URL ---------------------------------Registration
sip:luxapo-111@188.40.65.148 ---Fail
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| Thread: RE: Interoperability |
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Oh, I see, so YOU DID change that username setting on that trunk for me! Thank you very much! It is great. I did not think that this could have been a registration issue because
1) I was receiving calls, so I thought it was registered fine
2) I thought that my WRTP54G was also using this as it had "Use Auth ID" ticked:
see screenshot here: http://i48.tinypic.com/vmy6c2.png
The bottom line is that it is working great now, thanks a bunch!
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| Thread: RE: Interoperability |
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Hello Diafora,
thanks for getting back to me. I am very delighted to report that pbxes NOW WORKS perfectly, exactly as advertised. It is a little miraculous as I did not touch any settings, but it works. I don't think it was a registration problem as I was receiving calls fine. Looking at the logs the only difference I see is that in the IP column there is a "/sip.neophonex.hu" after my local IP, see these excerpts:
Successful call today:
Date Time Caller ID Number Destination IP Trunk Context App Duration
2010-05-25 11:07:36 "castro" <111> 0612222222 33.230-242-11.adsl-dyn.isp. belgacom.be/sip.neophonex.hu neophone from-internal-cont Dial 00:00:04
Unsuccessful calls earlier ("Your call could not be completed as dialed"):
Date Time Caller ID Number Destination IP Trunk Context App Duration
010-05-13 11:55:23 "castro" <111> 0613674359 212.135-240-11.adsl -dyn.isp.belgacom.be neophone from-internal-cont Hangup 00:00:00
2010-05-13 11:55:17 "castro" <111> 0613674359 212.135-240-11.adsl -dyn.isp.belgacom.be neophone from-internal-cont ResetCDR 00:00:05
Anyways, it works, and I am happy with it. Unfortunately I can not offer any advice to those having this same problem as I have no idea how it got fixed.
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| Thread: RE: Interoperability |
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Hi. I have set up two trunks and two extensions and it all went quite easily. I am able to intercomm, answer calls to either phonenumber and I could set up outbound routing as well. Only trouble is that I can not initiate calls on one of the trunks, I get a response like "Your call can not be completed as dialed. Please try your call again." This trunk is sip.neophonex.hu, whiich is on this compatibility list. When I connect with my ATA I can place calls without any trouble. I thought that maybe pbxes does not wait for the dialtone so I placed a W+. dial rule on this trunk to no avail. What am I doing wrong? Where should I look? This sip server should be on your list for a reason... TIA
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