[Asterisk-Users] Still no Broadvoice Outbound. (Bump)
Kris Edwards
krisedwards at gmail.com
Tue Mar 22 11:45:59 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-use
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