[Asterisk-Users] Last guy to get BV working outbound?
Kris Edwards
krisedwards at gmail.com
Mon Mar 21 19:11:24 MST 2005
I'm still not getting my outbound to work. I've seen two patches
relevant to broadvoice for chan_sip.c which apparently have already been
added to CVS. I'm dropping all outgoing calls after ~30 secs. Asterisk
doesn't seem to know they're gone though. I called my cell w/
broadvoice and turned on sip debug AFTer the call had physically dropped:
*CLI> sip show registry
Host Username Refresh State
sip.broadvoice.com:5060 310xxxMyBV at s 15 Registered
*CLI> dial 1509xxxMyCP
<< Console call has been answered >>
*Edit:This is irrelevant. I drop calls placed from a sip client
too/Edit. I can send/receive audio from the console*
ALSA lib pcm_hw.c:521:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy
ALSA lib pcm_hw.c:549:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed:
File descriptor in bad state
sip debug
SIP Debugging Enabled
*CLI>
<-- SIP read from 147.135.0.128:5060:
BYE sip:310xxxMyBV at 192.168.1.108 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qf87u30e8a009sk06s1.1sr
From:
<sip:1509xxxMyCP at sip.broadvoice.com>;tag=SD5g8g999-1221239059-1111455808838
To: "asterisk" <sip:310xxxMyBV at sip.broadvoice.com>;tag=as0bb17461
Call-ID: 480fa3c14d175b8a7eeeadc630af308e at sip.broadvoice.com
CSeq: 1 BYE
Content-Length: 0
Max-Forwards: 70
--- (8 headers 0 lines)---
<-- SIP read from 147.135.0.128:5060:
BYE sip:310xxxMyBV at 192.168.1.108 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qf87u30e8a009sk06s1.1sr
From:
<sip:1509xxxMyCP at sip.broadvoice.com>;tag=SD5g8g999-1221239059-1111455808838
To: "asterisk" <sip:310xxxMyBV at sip.broadvoice.com>;tag=as0bb17461
Call-ID: 480fa3c14d175b8a7eeeadc630af308e at sip.broadvoice.com
CSeq: 1 BYE
Content-Length: 0
Max-Forwards: 70
--- (8 headers 0 lines)---
<-- SIP read from 147.135.0.128:5060:
BYE sip:310xxxMyBV at 192.168.1.108 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qf87u30e8a009sk06s1.1sr
From:
<sip:1509xxxMyCP at sip.broadvoice.com>;tag=SD5g8g999-1221239059-1111455808838
To: "asterisk" <sip:310xxxMyBV at sip.broadvoice.com>;tag=as0bb17461
Call-ID: 480fa3c14d175b8a7eeeadc630af308e at sip.broadvoice.com
CSeq: 1 BYE
Content-Length: 0
Max-Forwards: 70
--- (8 headers 0 lines)---
Mar 21 20:56:22 NOTICE[29257]: chan_sip.c:4352 sip_reregister: --
Re-registration for 310xxxMyBV at sip.broadvoice.com@sip.broadvoice.com
11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.0.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK38aa5991
From: <sip:310xxxMyBV at sip.broadvoice.com>;tag=as61d09924
To: <sip:310xxxMyBV at sip.broadvoice.com>
Call-ID: 70a50d7620c7265410ed2cff3fb69d93 at sip.broadvoice.com
CSeq: 108 REGISTER
User-Agent: Asterisk PBX
Expires: 160
Contact: <sip:80171 at 192.168.1.108>
Event: registration
Content-Length: 0
---
<-- SIP read from 147.135.0.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.108:5060;received=165.166.232.49;branch=z9hG4bK38aa5991;rport=5060
From: <sip:310xxxMyBV at sip.broadvoice.com>;tag=as61d09924
To: <sip:310xxxMyBV at sip.broadvoice.com>;tag=SD500v699-
Call-ID: 70a50d7620c7265410ed2cff3fb69d93 at sip.broadvoice.com
CSeq: 108 REGISTER
Contact: <sip:80171 at 192.168.1.108>;expires=20
Content-Length: 0
--- (8 headers 0 lines)---
Mar 21 20:56:22 NOTICE[29257]: chan_sip.c:7659 handle_response: Outbound
Registration: Expiry for sip.broadvoice.com is 20 sec (Scheduling
reregistration in 15999 ms)
Destroying call '70a50d7620c7265410ed2cff3fb69d93 at sip.broadvoice.com'
<-- SIP read from 147.135.0.128:5060:
BYE sip:310xxxMyBV at 192.168.1.108 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qf87u30e8a009sk06s1.1sr
From:
<sip:1509xxxMyCP at sip.broadvoice.com>;tag=SD5g8g999-1221239059-1111455808838
To: "asterisk" <sip:310xxxMyBV at sip.broadvoice.com>;tag=as0bb17461
Call-ID: 480fa3c14d175b8a7eeeadc630af308e at sip.broadvoice.com
CSeq: 1 BYE
Content-Length: 0
Max-Forwards: 70
--- (8 headers 0 lines)---
<-- SIP read from 147.135.0.128:5060:
BYE sip:310xxxMyBV at 192.168.1.108 SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK2qf87u30e8a009sk06s1.1sr
From:
<sip:1509xxxMyCP at sip.broadvoice.com>;tag=SD5g8g999-1221239059-1111455808838
To: "asterisk" <sip:310xxxMyBV at sip.broadvoice.com>;tag=as0bb17461
Call-ID: 480fa3c14d175b8a7eeeadc630af308e at sip.broadvoice.com
CSeq: 1 BYE
Content-Length: 0
Max-Forwards: 70
--- (8 headers 0 lines)---
11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.0.128:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK60286504
From: "asterisk" <sip:asterisk at 192.168.1.108>;tag=as6d521e51
To: <sip:sip.broadvoice.com>
Contact: <sip:asterisk at 192.168.1.108>
Call-ID: 0a72198652f1d9677bbb1c19350ec6f9 at 192.168.1.108
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 22 Mar 2005 01:56:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
---
Mar 21 20:56:38 NOTICE[29257]: chan_sip.c:4352 sip_reregister: --
Re-registration for 310xxxMyBV at sip.broadvoice.com@sip.broadvoice.com
11 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.0.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK08f38249
From: <sip:310xxxMyBV at sip.broadvoice.com>;tag=as04d01a1c
To: <sip:310xxxMyBV at sip.broadvoice.com>
Call-ID: 70a50d7620c7265410ed2cff3fb69d93 at sip.broadvoice.com
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Expires: 160
Contact: <sip:80171 at 192.168.1.108>
Event: registration
Content-Length: 0
---
<-- SIP read from 147.135.0.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.108:5060;received=165.166.232.49;branch=z9hG4bK08f38249;rport=5060
From: <sip:310xxxMyBV at sip.broadvoice.com>;tag=as04d01a1c
To: <sip:310xxxMyBV at sip.broadvoice.com>;tag=SD500v699-
Call-ID: 70a50d7620c7265410ed2cff3fb69d93 at sip.broadvoice.com
CSeq: 109 REGISTER
Contact: <sip:80171 at 192.168.1.108>;expires=20
Content-Length: 0
--- (8 headers 0 lines)---
Mar 21 20:56:38 NOTICE[29257]: chan_sip.c:7659 handle_response: Outbound
Registration: Expiry for sip.broadvoice.com is 20 sec (Scheduling
reregistration in 15999 ms)
Destroying call '70a50d7620c7265410ed2cff3fb69d93 at sip.broadvoice.com'
Retransmitting #1 (no NAT) to 147.135.0.128:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:5060;branch=z9hG4bK60286504
From: "asterisk" <sip:asterisk at 192.168.1.108>;tag=as6d521e51
To: <sip:sip.broadvoice.com>
Contact: <sip:asterisk at 192.168.1.108>
Call-ID: 0a72198652f1d9677bbb1c19350ec6f9 at 192.168.1.108
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 22 Mar 2005 01:56:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
---
<-- SIP read from 147.135.0.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.108:5060;received=165.166.232.49;branch=z9hG4bK60286504;rport=5060
From: "asterisk" <sip:asterisk at 192.168.1.108>;tag=as6d521e51
To: <sip:sip.broadvoice.com>;tag=SD3giuc99-
Call-ID: 0a72198652f1d9677bbb1c19350ec6f9 at 192.168.1.108
CSeq: 102 OPTIONS
Accept: application/sdp,application/broadsoft,text/plain
Allow:
ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE,NOTIFY,UPDATE
Supported: 100rel,timer
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '0a72198652f1d9677bbb1c19350ec6f9 at 192.168.1.108'
<-- SIP read from 147.135.0.128:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.108:5060;received=165.166.232.49;branch=z9hG4bK60286504;rport=5060
From: "asterisk" <sip:asterisk at 192.168.1.108>;tag=as6d521e51
To: <sip:sip.broadvoice.com>;tag=SD3giuc99-
Call-ID: 0a72198652f1d9677bbb1c19350ec6f9 at 192.168.1.108
CSeq: 102 OPTIONS
Accept: application/sdp,application/broadsoft,text/plain
Allow:
ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE,NOTIFY,UPDATE
Supported: 100rel,timer
Content-Length: 0
--- (10 headers 0 lines)---
Destroying call '0a72198652f1d9677bbb1c19350ec6f9 at 192.168.1.108'
sip no debug
SIP Debugging Disabled
*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg
147.135.0.128 1509XXX57X 480fa3c14d1 00103/00103 ulaw Tx: ACK
1 active SIP channel(s)
*CLI> soft hangup SIP/sip.broadvoice.com-5604
Requested Hangup on channel 'SIP/sip.broadvoice.com-5604'
<< Hangup on console >>
*CLI> hangup
*CLI>
I'd post my sip.conf, but it's pretty much configured as it is in the
wiki (with the exception of user=phone??) Anyway, I am using NAT. I
tried DMZ w/ NAT off but it made no difference (I have no way to get a
true external IP. DMZ for me is port forwarding, but I had the same
results).
Any suggestion would be aprreciated.
Thanks!
kRis
Brian G wrote:
> Rich thanks, this makes it a little clearer. My servers are using NAT
> behind a Cisco PIX. I only needed the simple patch (see below).
> I configured sip.conf from these instructions:
>
> http://www.voip-info.org/wiki-Asterisk+settings+Broadvoice
>
> Hope this helps somebody. Sorry I wasn't clear about using NAT.
>
> Brian
>
> Patch I used:
>
> --- chan_sip.c.fcs 2003-12-13 14:54:37.000000000 -0800
> +++ chan_sip.c 2005-03-10 11:48:40.000000000 -0800
> @@ -4444,10 +4446,10 @@
> }
>
> static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req,
> char *header, char *respheader, char *msg, int init) {
> - char digest[256];
> + char digest[1024];
> p->authtries++;
> memset(digest,0,sizeof(digest));
> - if (reply_digest(p,req, "Proxy-Authenticate", msg, digest,
> sizeof(digest) )) {
> + if (reply_digest(p,req, header, msg, digest, sizeof(digest) )) {
> /* No way to authenticate */
> return -1;
> }
>
>
> On Sat, 2005-03-19 at 09:14, Rich Adamson wrote:
>
>>A lot of the BV config confusion is the result of users with registered
>>IP's vs nat'ed IPs. The patch _was_ only required for those that used
>>nat'ed systems (proven shortly after that patch was released, and backed
>>by those that wrote the patch).
>>
>>So, for those that are still mucking around with BV configs, it would
>>be helpful to others on this list to understand whether your systems are
>>nat'ed or not in initial posts.
>>
>>You can also help yourself by validating some of these recommended
>>parameters against those listed in /usr/src/asterisk/configs/sip.conf.samples.
>>(User=phone is one such example of a do-nothing statement that has
>>no meaning whatsoever.)
>>
>>Since I no longer subscribe to BV's service, I don't have a clue
>>which * releases need the patch and which don't.
>>
>>------------------------
>>
>>
>>>Thanks John, but I tried adding those and many others. Turned out that
>>>I needed to install a patch even though I tried CVS-3/11/05 and
>>>CVS-3/17/05 code. I'm not sure what release needs what patch to work
>>>but I definitely needed a patch. Thanks to the person on this list who
>>>sent it along. There are many people with many configs posting on many
>>>lists but I can't say I have a handle it.
>>>
>>>Brian
>>>
>>>On Fri, 2005-03-18 at 12:30, John Sawa wrote:
>>>
>>>>Brian,
>>>>
>>>>You will need to add the following to your broadvoice peer:
>>>>
>>>>user=phone
>>>>insecure=very
>>>>dtmf=inband
>>>>
>>>>For more info check out:
>>>>
>>>>http://geekgazette.com/index.php?option=com_content&task=view&id=20&Itemid=26
>>>>
>>>>Hope this helps. -john
>>>>
>>>>
>>>>Brian G wrote:
>>>>
>>>>
>>>>>I have tried everything to get BV working outbound. All worked fine
>>>>>until the BV change last week. I called BV and they changed me to sip
>>>>>gen with a new password. I stripped my Asterisk server to one phone on
>>>>>Zap/1 until I get this working. The same BV account works fine with a
>>>>>SPA-3000 so I don't suspect a firewall problem.
>>>>>
>>>>>Symptoms: Asterisk registers with BV Ok
>>>>>Incoming calls work
>>>>>Outbound calls send Invite, receive 100, then 401
>>>>>Asterisk sends an ACK instead of another Invite with credentials
>>>>>
>>>>>If anyone knows what specifically makes Asterisk respond to the 401 with
>>>>>credentials for an authenticated Invite, I'd appreciate it. I can't
>>>>>seem to find this out.
>>>>>
>>>>>Thanks in advance,
>>>>>Brian
>>>>>
>>>>>Here is my sip.conf:
>>>>>
>>>>>[general]
>>>>>port = 5060 ; Port to bind to
>>>>>bindaddr = 0.0.0.0 ; Address to bind SIP channel to
>>>>>context = default ; Default context for incoming calls
>>>>>srvlookup = yes ; Enable DNS SRV lookups on outbound
>>>>>calls
>>>>>
>>>>>disallow=all ; Disallow all codecs
>>>>>allow=ulaw ; Allow codecs in order of preference
>>>>>;
>>>>>; Configuration for BroadVoice
>>>>>;
>>>>>register =>
>>>>>508XXXXXXX at sip.broadvoice.com:pword:508XXXXXXX at sip.broadvoice.com
>>>>>;
>>>>>[broadvoice]
>>>>>type=peer
>>>>>host=sip.broadvoice.com
>>>>>secret=pword
>>>>>fromuser=508XXXXXXX
>>>>>username=508XXXXXXX
>>>>>authuser=508XXXXXXX
>>>>>fromdomain=sip.broadvoice.com
>>>>>context=incoming
>>>>>canreinvite=no
>>>>>dtmfmode=inband
>>>>>qualify=yes
>>>>>
>>>>>in extensions.conf:
>>>>>[default]
>>>>>exten => _81XXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@broadvoice)
>>>>>exten => _81XXXXXXXXXX,2,Congestion()
>>>>>exten => _81XXXXXXXXXX,102,busy()
>>>>>
>>>>>Other Asterisk info:
>>>>>
>>>>>*CLI> sip show registry
>>>>>Host Username Refresh State
>>>>>147.135.0.128:5060 508XXXXXXX 120 Registered
>>>>>*CLI>
>>>>>*CLI> show version
>>>>>Asterisk CVS-03/11/05-16:07:49 built by root at hostname.com on a i686
>>>>>running Linux
>>>>>*CLI>
>>>>>*CLI> Mar 17 10:35:08 NOTICE[-245486672]: chan_sip.c:5047
>>>>>handle_response: Failed to authenticate on INVITE to '"Analog1"
>>>>><sip:508XXXXXXX at sip.broadvoice.com>;tag=as212bf17
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>_______________________________________________
>>>>>Asterisk-Users mailing list
>>>>>Asterisk-Users at 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 mailing list
>>>>Asterisk-Users at 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
>>>
>>>_______________________________________________
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>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>---------------End of Original Message-----------------
>>
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>
>
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