[asterisk-users] sip trunk, parsing DID
Hello,
I am using a Swiss VoIP provider called sipcall. They have what they
call a SIP trunk, and it is less expensive than individual accounts. From
Asterisk's point of view, this is just a regular SIP account, which
can however receive and send calls from multiple numbers. I just migrated
from individual SIP accounts terminated on my Asterisk to one single
SIP trunk.
It works perfectly (in and out). For outgoing calls, it's just
sufficient to set CALLERID(num) to the appropriate number you want the
call to originate from (easy!).
For incoming calls, here is an example SIP message, with MY_IP, SIPCALL_IP,
DEST_NUMBER AND SRC_NUMBER replacing the actual values:
INVITE sip:s@MY_IP:5060 SIP/2.0
Via: SIP/2.0/UDP SIPCALL_IP:5060;branch=z9hG4bK3ee1k92090iihapdm420.1
Max-Forwards: 67
Contact:
To:
From: ;tag=hy4fwr752woo42uj.o
Call-ID: 1663976908-326811297@1~1o
CSeq: 867 INVITE
Expires: 300
Allow: INVITE, ACK, BYE, CANCEL, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE,
OPTIONS, UPDATE
Content-Disposition: session
Content-Type: application/sdp
User-Agent: PortaSIP
h323-conf-id: 3912070954-288423879-3678105731-1479596074
cisco-GUID: 3912070954-288423879-3678105731-1479596074
Content-Length: 262
Since it looks that only the To: header contains the real destination number,
and debugging shows that it is not copied in ${CALLERID(all)} nor ${EXTEN}, I
had to revert to this hack, which works great:
exten => s,1,Log(NOTICE, Incoming call from sipcall-trunk ${CALLERID(all)}
to ${EXTEN} DID ${SIP_HEADER(To)})
exten => s,n,Set(DID=${SIP_HEADER(To):})
exten => s,n,Set(DID=${DID:5:11})
exten => s,n,Log(NOTICE, Parsed DID: ${DID})
exten => s,n,Goto(sipcall-trunk,sipcall-${DID},1)
exten => s,n,Hangup()
I then have individual sipcall-NUMBER handling the actions for the individual
numbers.
Is there a simpler way? Is there a safer way (check that DID only contains
numbers, e.g.?)
Thank you for any ideas or pointers.
--
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Re: [Asterisk-Users] Spoofing a BLF Signal?
On Wed, May 24, 2006 at 08:54:09PM -0400, Matt wrote: Then bristuff may be the way to go. However, I read this on the wiki Note: Using bristuff breaks PRI support, so you cant have both bri and pri in the same server. That's not good! I need PRI. I never had this problem, using a BRI and a PRI on the same machine with bristuff patches. Ok, a quite old version of them, though. The only side effet of bristuff is that it will prevent you to link proprietary (non GPL) driver into it and redistribute. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to force Requested transfer capability on BRI/PRI dial?
Hi,
on a configuration with one external ISDN S bus (to telco) and one
internal S bus (to ISDN telephone), where Asterisk is in the middle
(using HFC hardware), I noted the following:
- when a GSM phone or ISDN phone calls in, the Transfer capability
is Requested transfer capability: 0x00 - SPEECH
- when an analog phone calls in (either from an analog line or
an analog ISDN port), the Transfer capability is
Requested transfer capability: 0x10 - 3K1AUDIO
The problem is that the ISDN phone in the internal bus apparently
ignores this transfer capability in the SETUP message: it does not
answer at all. It works well in the first case (GSM, etc)
AFAIK there is no difference in the codec in both cases: just a type
difference. How can I make Asterisk (libpri?) force the Transfer
capability to the first value in any case ?
After looking a bit in the source (channels/chan_zap.c) and in
apps/app_settransfercapability.c it looks like I can do it
with in the call sequence (extensions.conf)
SetTransferCapability('SPEECH')
Is there a better way ?
Thank you.
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Re: [Asterisk-Users] PortaSIP/PortaBilling incompatibility (provider: sipcall.ch)
On Mon, Apr 11, 2005 at 08:18:43PM +0200, gramels wrote: If Useragent field in this config corresponds to User-Agent field in Asterisk's SIP messages and you may change it to something that doesn't contain a word Asterisk - please try to do so; in such case PortaSIP will not apply remote IP auth. I might have a similar problem with SER (sipphone.com) and my Asterisk. However the mentionned work around doesn't work. Funnily the register works with the same password. What happens: Apr 13 09:34:45 NOTICE[2495]: chan_sip.c:6831 handle_response: Failed to authenticate on INVITE to '17476691152 sip:[EMAIL PROTECTED];tag=as41277c10' log: Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK7e66a7cc From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: portasipfriendly Date: Wed, 13 Apr 2005 07:34:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 343 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK7e66a7cc To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. WWW-Authenticate: Digest realm=sipphone.com, nonce=425cc84ac3c477a344ab166ec9 Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=1706 req_src_ip=80.83.46.147 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri= Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK7e66a7cc From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e.1876 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: portasipfriendly Content-Length: 0 Reliably Transmitting: Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK6ace6db5 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: portasipfriendly Authorization: Digest username=17476691152, realm=sipphone.com, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=425cc84ac3c477a344ab166ec9 7f6efdadcc6bb3, response=1bdef16f3de89e9194116a2a0135a495, opaque= Date: Wed, 13 Apr 2005 07:34:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 343 Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK6ace6db5 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. bd57 Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE WWW-Authenticate: Digest realm=sipphone.com, nonce=425cc84bb62fad894c1533475c 08dead3e27baa5 Content-Length: 0 Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=1707 req_src_ip=80.83.46.147 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri= sip:[EMAIL PROTECTED] via_cnt==1 Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK6ace6db5 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. bd57 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 ACK User-Agent: portasipfriendly Content-Length: 0 Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.83.46.147:5060;branch=z9hG4bK02c48610 From: 17476691152 sip:[EMAIL PROTECTED];tag=as41277c10 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE User-Agent: portasipfriendly Authorization: Digest username=17476691152, realm=sipphone.com, algorithm=MD 5, uri=sip:[EMAIL PROTECTED], nonce=425cc84bb62fad894c1533475c 08dead3e27baa5, response=219b7fec33546a32a830edfba25fa601, opaque= Date: Wed, 13 Apr 2005 07:34:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Sip read: SIP/2.0 401 Unauthorized To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. WWW-Authenticate: Digest realm=sipphone.com, nonce=425cc84bb62fad894c1533475c Warning: 392 198.65.166.131:5060 Noisy feedback tells: pid=1702 req_src_ip=80. 83.46.147 req_src_port=5060 in_uri=sip:[EMAIL PROTECTED] out_uri= Transmitting: To: sip:[EMAIL PROTECTED];tag=21a483426c2cd5d9b85bffe6bba40a2e. 6caf Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 ACK User-Agent: portasipfriendly Content-Length: 0 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] Reject second IAX call
Hi, is there a configuration in iax.conf to specify that if a call goes to that peer, a second call should not be allowed. Specifically, I do this: Dial(IAX2/iaxcomm) # in extensions.conf for a specific extension in iax.conf: [iaxcomm] type=friend mailbox=20 accountcode=iaxcomm username=iaxcomm host=dynamic auth=md5,plaintext,rsa secret=fksjdfh73 ; changed context=local-iaxcomm permit=192.168.10.0/24 allow=ulaw is there an option to disable a 2nd call? thank you. PS: the real problem in my case is that for some reason IAXcomm sees a second call coming in after 30 sec - 1 minute on 2 over 10 incoming calls. This phantom call must be disconnected to resume the real call. Funny duh? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Thu, Mar 24, 2005 at 05:36:35PM +0200, Mark Elkins wrote: I am still curious. Which Driver do you use for the HFC card? oh, specifically I use: zaphfc_0.2.0-RC7j_florz-4.diff.gz applied on top of RC7k ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Thu, Mar 24, 2005 at 05:36:35PM +0200, Mark Elkins wrote: I am still curious. Which Driver do you use for the HFC card? I manage my own Debian package repository for Debian stable (woody) backports of asterisk and related stuff (based somewhat on backports.org). I currently use: Version: 1:1.0.6-0.bristuff_0.2.0_RC7k.cril.0 so it's about the same (I just have a few additional patches). It could be: bristuff-0.2.0-RC7k stuff from http://www.junghanns.net/ - but this locks you into using a particular - non-HEAD version of Asterisk.. (and missing all the new goodies) not really, AFAIK the last time I tried, the BRI patches apply cleanly also to more recent versions. And 1.0.6 works quite well for me. The only problem I still have with 1.0.6 is that for some reason, IAXcomm user tell me that when they get a call, and answer it, then after 30 seconds or 1 minute a new call comes in (which is fake) and you have to cancel it in IAXcomm to get the first call correctly. I haven't debugged it yet. I have SIP phones, IAX2 connections to remote Asterisk, ISDN bidirectionnal gateway, analog TDM board with el cheapo analog tel, DECT CLIP-compatible phone (works), and also ISDN local phones (using HFC NT mode). I wish there were single, four and eight port ISDN BRI cards that Digium sold and supported - so I could run whichever version of Asterisk I wanted... ISDN was never popular in the US for BRI lines. In Europe we even do stupid things such as multiple-BRI operated in cascade (e.g. 4 BRIs, giving you the equivalent of 8 communication channels), where it would be more intelligent to use (partial) E1 for that purpose. I think Germany has those partial E1 available to the public. In the US, people usually do analog upto 10 lines and then get a T1. As analog lines include caller ID (however AFAIK no easy ability to *set* outgoing caller ID nor real calle*d* ID, without distinctive ringing), most benefits of BRI ISDN are unneeded. That's why most BRI ISDN development is done in Europe -- or more precizely looks like it's Germany, really. An alternative to zaptel is to use the m_isdn implementation of the Linux kernel. As I use 2.4 and it works very well with zaptel/zaphfc, I didn't bother to try the 2.6 (crappy) kernels or the 2.4-m_isdn backport yet. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Wed, Mar 23, 2005 at 06:55:28PM +0200, Mark Elkins wrote: Last time I tried - there were a few problems... I had a few random crashes, higher delays and echo with the EICON. I replaced it now with an HFC. The EICON on isdn4linux was however a bit better than the AVM C4 with CAPI. 1 - Outbound DTMF - never made it... ie You can not interact with someone else's IVR (DTMF controlled systems) This is because DTMF detection and sending is disabled in chan_modem. Use the information from http://www.marko.net/asterisk/archives/0301/0849.html. Also, this is because there was/is an issue with Asterisk and modem/i4l DTMF support and config. modem.conf: dtmfmode=asterisk 2 - Inbound DTMF - Certain voices would be interpreted as DTMF - which is fine until they sounded like a '#' - and got transfered (some strange reason - my wife's voice - especially when she got angry) This is because the kernel driver thought it should do detection and does it badly. Apply patch http://lists.digium.com/pipermail/asterisk-users/2003-June/014104.html to kernel. In my case interaction with remote IVR and inbound DTMF was working perfectly after this and no more beeps where received during the phone call. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC/zaphfc/zaptel: issues with multiple inbound calls
Hi, with Asterisk 1.0.1 and bri-stuff-0.1.0-RC4a, and two calls already established on the ISDN BRI, the third call causes scratches in already running calls and an answer of an unexisting channel: Ring on unconfigured channel 0/0 span 2 with Asterisk 1.0.5 and bristuff-0.2.0-RC5, this bug is fixed. The third call produces a: Ignoring callwaiting SETUP on channel 0/0 span 2 0 this is good. However, it has the side effect of preventing any second call on the BRI! In short, once a third call came in, there can be only upto one inbound call. Did someone also experiment this behaviour ? thank you. (HFC-s) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange MSN issue with HFC-s
Hi, I have two HFC-s boards I configured in NT and TE mode respectively. When I connect the two boards together, I can dial extensions and I see the correct called and caller ID numbers: -- Executing SetCallerID(Zap/2-1, 7516862) in new stack == CDR updated on Zap/2-1 -- Executing Dial(Zap/2-1, Zap/g2/0795025602|30|r) in new stack -- Called g2/0795025602 -- Extension '0795025602' in context 'isdn-local-bus' from '7516862' does not exist. Rejecting call on channel 0/1, span 1 however, when I connect the TE card to my NT2ab, it seems the caller ID number is not passed correctly (?) to the telco, since if when I get the call on my mobile I get the main number for my ISDN connection, not the specified number. With an AVM c4 it works correctly (syntax is CAPI/FROM:TO). Thank you for any help! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange MSN issue with HFC-s
Hm, do you have the right settings in zapata.conf? (switchtype, pridialplan...) So, in Switzerland, I assume switchtype = euroisdn now, for the pridialplan, am I right that the pridialplan configures the way the phone number to be dialed (called ID) is sent, and that the prilocaldialplan denotes the way the caller ID is sent to the telco ? The fact is that anything else than pridialplan = local and prilocaldialplan = local prevent any dialing out. I haven't yet understood how this impacts the way the ID is sent out. At least on CAPI, I need to set the MSN without the prefix (e.g. 7516862 and not 0327516862). This is what I tried. Also, am I right that `callerid=asreceived' tells the received caller ID to Asterisk when a call comes in ? And has nothing to do with the way caller ID is sent to the telco ? [ BRI interface with HFC-s in TE mode ] probably I will need to add the patches for the ISDN analyzer to see what happens. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange MSN issue with HFC-s
As a complement: probably I will need to add the patches for the ISDN analyzer to see what happens. - when an ISDN phone calls out and the MSN is correctly passed: Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '0327516862' ] - when the HFC-s dials out: Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number passed network screening (1) '0327516862' ] - when the network calls: Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '0328414004' ] So for me it looks like the problem could be fixed by making the HFC-s change the TON from 4 to 0 and the presentation from 1 to 0. However so far I have no clue how to do that. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Strange MSN issue with HFC-s
On Thu, Feb 17, 2005 at 07:13:38PM +0100, Marc SCHAEFER wrote: As a complement: and the fix: prilocaldialplan=unknown thanks to the people here and neolynx. This fixes the problem with the caller ID not set with an HFC-s due to bad Type of Number in the ISDN SETUP message (at least for my telco). [ and sorry for the rate of messages ] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Probs with oh323 driver: audio only in 1 direction
On Tue, May 04, 2004 at 09:30:01AM +0200, Michael Niehren wrote: try to setup asterisk as an ISDN2H323-Gateway. The only problem i have after establishing a call is, that Audio works only from IP to ISDN-Phone but not from ISDN to IP-Phone. I can add that it works perfectly from a SIP phone to H.323, but not from ISDN to H.323. (and it works from ISDN to SIP phone). My symptom is the same: you can hear H.323 from ISDN, but not hear ISDN from H.323. However I use H.323 from asterisk's recent CVS, not oh323. I use codec GSM in h323.conf. Logs excerpt: Failure: (from ISDN) May 5 09:10:21 DEBUG[23573]: channel.c:1331 ast_indicate: Driver for channel 'CAPI[contr4/8414774]/1' does not support indication 3, emulating it -- H323/1.2.3.4 answered CAPI[contr4/8414774]/1 May 5 09:10:21 DEBUG[23573]: rtp.c:1098 ast_rtp_write: Ooh, format changed from UNKN to GSM Ok: (from SIP) May 5 09:10:37 DEBUG[25621]: rtp.c:1098 ast_rtp_write: Ooh, format changed from UNKN to ULAW May 5 09:10:40 DEBUG[25621]: channel.c:1331 ast_indicate: Driver for channel 'SIP/17476691152-9ccd' does not support indication 3, emulating it May 5 09:10:41 DEBUG[25621]: rtp.c:1098 ast_rtp_write: Ooh, format changed from UNKN to GSM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Discriminate on IAXTEL dial-in
On Sun, Mar 21, 2004 at 07:42:05AM -0600, Eric Wieling wrote: You are correct, IAXtel does not send the called number. Calls from both IAXTel accounts will fall into the s extension. Oh, I see. So I have now implemented a menu. If you call 1-700-895-5211 you can now dial 0800 numbers in Switzerland (dial 00800 800 800 for Swisscom fixnet for example). If you dial +41 328 41 47 74 you get the other way around, ie dial into IAX. And I am extension 200 (or 9). I however have another question: - apparently when I call from ISDN to an IAX gnophone, I get a very short ring then an error: (XXX are mine) -- Calling using options 'exten=s;callerid=03284140XX;language=en;formats=2;capability=65283;version=1;adsicpe=0' -- Called XXX -- Call accepted by 80.83.50.XXX (format GSM) -- Format for call is GSM -- IAX[XXX]/50 is ringing Mar 21 20:43:29 DEBUG[33810]: channel.c:1265 ast_indicate: Driver for channel 'CAPI[contr4/8414774]/10' does not support indication 3, emulating it Mar 21 20:43:29 ERROR[33810]: chan_capi.c:851 capi_write: not a voice frame Mar 21 20:43:29 WARNING[33810]: app_dial.c:313 wait_for_answer: Unable to forward image Mar 21 20:43:29 DEBUG[33810]: chan_iax.c:1861 iax_hangup: We're hanging up IAX[XXX]/50 now... -- Hungup 'IAX[XXX]/50' == Spawn extension (macro-dial-extension, s, 3) exited non-zero on 'CAPI[contr4/8414774]/10' in macro 'dial-extension' - the problem doesn't happen when calling from a SIP phone. - the problem also happens if you do ISDN - IAX - IAX gnoèphone. Probably this is a bug in chan_capi-0.3.0 ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Discriminate on IAXTEL dial-in
- apparently when I call from ISDN to an IAX gnophone, I get a very short ring then an error: (XXX are mine) This doesn't happen when gnophone is configured as `Use Asterisk' apparently. So this is now solved. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Experimental Switzerland - IAX gateway
Hi, to test my Asterisk / IAX connection I have configured the Swiss phone number 032 841 47 74 to a IAX gateway. You can dial 1-700, 1-800 and other numbers from this number (prefix with 00: for example 0018005551212). This is a local rate number. I have not yet implemented IAXtel - Swiss 1-800 yet because I didn't succeed in registering two IAXtel numbers yet. Feel free to test this, for example to test dialing into your gnomephone application. It may be stopped at any time, will probably work mostly week-end and working hours GMT+1 at this time. This test might however be discontinued at any time. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] double-dial in SIP Grandstream
Hi, I have even now connected to IAXtel at number 1-700-895-5211 when I am in the office, so Asterisk is great. I just found something strange, which is that if I am already in a connection with my Grandstream and talking, and a second call comes in, it rings on the Grandstream. However, if I am not talking but waiting for dialing, the caller gets a busy signal (good). How can I make sure there is only one call at a time to the SIP phone ? (call waiting could be useful, but I didn't figure out how to do this with the SIP). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN debugging and SIP dial-in issue]
(I have some problems with my mailing-list alias, I hope this doesn't get sent twice) On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote: Thank you for your comments Philipp: - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 That doesn't look right. Do you have another SIP server installed on your client machine - shouldn't that rather be *, or did you - which I guess - just mistype the IP? Which SIP phone are you using Mis-typed, yes. The SIP server is the Asterisk server and is 192.168.1.10 (hardware/software, brand, version)? Grandstream BudgeTone-100 - can dial 1-800-CALL-ATT and talk with an operator through the sipphone.com SIP proxy, quality is adequate (changed the SIP server to sip01.sipphone.com of course) - when the SIP server is Asterisk, can be dialed from ISDN without any problem (maybe a slight delay), quality is good both directions. - can dial to Asterisk, in that case Asterisk's debug shows the call, but fails. Nothing is hearable on the BudgetTone except a busy tone. Software: Program--1.0.3.81Bootloader--1.0.0.7HTML--1.0.0.18 Call examples: (this time with `sip debug' I just found about) SIP phone dials '2' Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.190 From: Martial Guex sip:[EMAIL PROTECTED];tag=7adc221a-d23b-5289-93ff-261810e5291c To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 53320 INVITE User-Agent: Grandstream SIP UA 1.0.3.81 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 266 [ ... ] Sending to 192.168.1.190 : 5060 (non-NAT) [ ... ] Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 DEBUG[5126]: File chan_sip.c, Line 3965 (check_user): Setting NAT on RTP to 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required [ ... ] ACK sip:[EMAIL PROTECTED] SIP/2.0 [ ... ] DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 53320: Found [ ... ] DEBUG[5126]: File chan_sip.c, Line 991 (find_user): Call from user '17476691152' is 1 out of 0 Looking for 2 in localphones DEBUG[5126]: File chan_sip.c, Line 3369 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED] -- Executing Playback(SIP/17476691152-a52e, publicar-extbusy|skip) in new stack *CLI some time ... a few seconds No such command 'some' (type 'help' for help) *CLI -- Timeout on SIP/17476691152-a52e == CDR updated on SIP/17476691152-a52e -- Executing Hangup(SIP/17476691152-a52e, ) in new stack == Spawn extension (localphones, t, 1) exited non-zero on 'SIP/17476691152-a52e' DEBUG[15376]: File chan_sip.c, Line 1068 (sip_hangup): find_user(17476691152) - decrement inUse counter Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden - dial-in from ISDN, then transfer to ISDN on the secondary channel: doesn't work (more details below) I assume with transfer you mean that you are trying to dial out on the 2nd channel. So who are you trying to call? If you are trying to call I call from a mobile phone to a mobile phone: mobile - ISDN in - ISDN out - mobile this setup works with software I developped (a modified isdn2h323 which can connect the two streams by byte-copying, plus conferencing and control software). Not sure, but: You might want to look into the isdn4linux documentation and use its tools like isdnlog (?) etc. I added some printf()'s in channels/*modems*.c and the adequate AT commands are sent, something wrong is happening but it's not Asterisk's fault. If that is not it: Check your context setup: The incoming call must be in a context that is allowed to dial out again. There is no immediate error, looking like some attempt is made. Please provide (the relevant parts of) your extensions.conf. [xfertomobile] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,Background(transfer); schaefer exten = s,4,Dial,Modem/g1:079xxx|60|r exten = s,5,Playback(extbusy,skip) ; schaefer exten = s,6,Hangup ; schaefer [localphones] exten = s,1,Wait,1 ; Wait a second, just for fun exten = s,2,Answer ; Answer the line exten = s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = 1,1,Goto(demo,s,1) exten = 2,1,Playback(extbusy,skip) exten = 3,1,Goto(xfertomobile,s,1) exten = t,1,Hangup exten = i,1,Playback(invalid) ; That's not valid, try again [default] include = xfertomobile - check rtp.conf I will need help here. Configuration on the SIP phone is local port 5004 and don't use random
Re: [Asterisk-Users] ISDN debugging and SIP dial-in issue]
On Mon, Nov 17, 2003 at 02:08:30PM +0100, Philipp von Klitzing wrote:
You should also add to sip.conf for [17476691152]:
disallow=all
allow=ulaw
allow=alaw
This was the key. I now hear the voice prompts correctly.
rather use the new syntax for the Dial application like
Dial(Modem/g1/012345,20,rt).
I now use the CAPI driver, which works fine, including
call transfer by copying.
My dial plan:
; Dial plan:
; 0 001 747 xxx routed to sipphone.com with 1152's registration
; 0 001 800 xxx routed to sipphone.com with 1152's registration
; 0 x* routed to ISDN dialup (allowed from local SIP only)
; 100 the demo
; NOTES
;- Calls from sipphone.com go to default, which is xfertomarclocal (SIP)
;- isdn-dial up can also dial to sipphone.com or go to the demo through
; an escape.
My configurations:
- dial in from ISDN goes first to SIP phone then to external
mobile phone through ISDN call copying:
(could also be done through not answering and deflecting)
[isdn-in]
include = no-suckers
include = xfertomarc
- local SIP phone dials out with '0' on ISDN. Special cases go to
sipphone.com
[local-sip-in]
include = sipphone.com
include = isdn-dial-out ; order matters.
exten = 100,1,Goto(demo,s,1)
[xfertomarc]
include = xfertomarclocal
exten = s,3,Goto(xfertomobile,s,4) ; ---
[xfertomarclocal]
exten = s,1,Answer ; Probably could also defer answering
exten = s,2,Dial(sip/17476691152,30) ; schaefer
[xfertomobile]
exten = s,1,Wait,1 ; Wait a second, just for fun
exten = s,2,Answer ; Answer the line
;exten = s,3,Playback(transfer,skip); schaefer
exten = s,3,Background(transfer); schaefer
exten = s,4,Dial,CAPI/8414014:x|30|r
exten = s,5,Playback(publicar-extbusy,skip) ; schaefer
exten = s,6,Hangup ; schaefer
[sipphone.com]
exten = s,1,Answer ; dummy for Goto
exten = s,2,DigitTimeout,10
exten = s,3,ResponseTimeout,10
exten = i,1,Playback(invalid) ; That's not valid, try again
exten = _001747NXX,1,SetCallerID(${CALLERIDNUM})
exten = _001747NXX,2,SetCIDName(${CALLERIDNUM})
exten = _001747NXX,3,Dial(Sip/${EXTEN:[EMAIL PROTECTED])
exten = _001747NXX,4,Playback(invalid)
;exten = _001747NXX,5,Hangup
exten = _001800NXX,1,SetCallerID(${CALLERIDNUM})
exten = _001800NXX,2,SetCIDName(${CALLERIDNUM})
exten = _001800NXX,3,Dial(Sip/${EXTEN:[EMAIL PROTECTED])
exten = _001800NXX,4,Playback(invalid)
;exten = _001800NXX,5,Hangup
Thank you for your help.
I still have one -- probably NAT related -- issue with
contacting SIPphone.com (no sound). Will document it.
___
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[Asterisk-Users] ISDN debugging and SIP dial-in issue
Hi, my setup is quite simple: an asterix CVS of 2003-11-15 on a 2.4.21-debian-5 GNU/Linux box in an internal network (192.168.1.0/24, asterisk is 192.168.1.10). - with a SIP phone configured as 192.168.1.190, and with its SIP server being 192.168.1.190 - with an ISDN AVM c4 i4l card on an ISDN connection with 2 channels. I try to: - dial-in from ISDN, then transfer to the SIP phone: works very well. - dial-in from ISDN, then transfer to ISDN on the secondary channel: doesn't work (more details below) - dial anything from the SIP phone: doesn't work (more details below) the very good Asterisk basic demos (echo, IAX) work very well. Details: - ISDN dial-out: -- Executing Dial(Modem[i4l]/ttyI0, Modem/g1:079xxx|60|r) in new stack DEBUG[15376]: File app_dial.c, Line 392 (dial_exec): SIMPLE DIAL (NO URL) -- Called g1:079xxx (xxx are from me) - SIP dial in: it seems the session is initiated (SIP message from Asterisk on the ethernet), and then UDP (voice?) packets are sent, but no answer comes from the SIP phone and after a moment Asterisk fails with: DEBUG[5126]: File chan_sip.c, Line 3369 (build_route): build_route: Contact hop: sip:[EMAIL PROTECTED] -- Executing Playback(SIP/17476691152-7158, extbusy|skip) in new stack -- Timeout on SIP/17476691152-7158 == CDR updated on SIP/17476691152-7158 -- Executing Hangup(SIP/17476691152-7158, ) in new stack == Spawn extension (localphones, t, 1) exited non-zero on 'SIP/17476691152-7158' DEBUG[15376]: File chan_sip.c, Line 1068 (sip_hangup): find_user(17476691152) - decrement inUse counter DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Response 33505: Found My specific questions are: - how can I see the i4l chatting dialing-out to be sure what the problem is (could be a wrong MSN for example, or Asterisk interpreting the 0 prefix) - what should I do for this SIP dial-in issue ? Specifically how can I debug this ? Thank you very much for any ideas/pointers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
