[Asterisk-Users] Can't bridge between h323 and sip calls
Ronald Wiplinger
ronald at elmit.com
Thu Jun 30 08:44:09 MST 2005
Alex Vishnev wrote:
>Hello,
>
>I am using asterisk CVS-head from 6/28. I am also using chan_oh323 that
>comes with asterisk. I tried to place a call from h323 device into asterisk.
>in extensions.conf, I routed the call to my sip phone. The sip phone was
>already registered with asterisk. all the signaling looks ok to me. The sip
>phone rings when h323 call hits the asterisk box. But then the call is
>dropped. It appears that asterisk is trying to convert incoming g.729 codec
>to ulaw and it can't. I was assumed that g.729 will pass-thru to the phone.
>In fact, when an invite is sent bothg G729, G723 are codecs in SDP. However,
>when SIP phone answers, it only replies with g723 on 200OK. I am still
>unclear about that, but that's not really that important. I would like to
>find out why I can't bridge these two legs. below is the trace from the
>call. I am suspecting that a line below is the cause, but not sure why. Can
>someone help???
>
>
Can you show your settings? extensions.conf (for that part) and h323.conf
bye
Ronald
>Jun 29 10:59:46 WARNING[8862]: app_dial.c:1324 dial_exec_full: Had to drop
>call because I couldn't make H323/ip$64.243.115.153:32971/11679 compatible
>with SIP/debit-9f37
>
>-----asterisk log------
>
> -- Executing Dial("H323/ip$64.243.115.153:32971/11679",
>"SIP/debit|20|rt") in new stack
>Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
>path from g729 to ulaw
>Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
>path from g729 to ulaw
>We're at 64.243.115.157 port 18192
>Answering with capability 0x1 (g723)
>Answering with capability 0x4 (ulaw)
>Answering with capability 0x8 (alaw)
>Answering with capability 0x100 (g729)
>Answering with non-codec capability 0x1 (telephone-event)
>12 headers, 13 lines
>Reliably Transmitting (NAT) to 69.115.205.168:4152:
>INVITE sip:debit at 69.115.205.168:4146 SIP/2.0
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>To: <sip:debit at 69.115.205.168:4146>
>Contact: <sip:7323600296 at 64.243.115.157>
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 102 INVITE
>User-Agent: Asterisk PBX
>Date: Wed, 29 Jun 2005 14:59:41 GMT
>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
>Content-Type: application/sdp
>Content-Length: 292
>
>v=0
>o=root 8862 8862 IN IP4 64.243.115.157
>s=session
>c=IN IP4 64.243.115.157
>t=0 0
>m=audio 18192 RTP/AVP 4 0 8 18 101
>a=rtpmap:4 G723/8000
>a=rtpmap:0 PCMU/8000
>a=rtpmap:8 PCMA/8000
>a=rtpmap:18 G729/8000
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-16
>a=silenceSupp:off - - - -
>
>---
> -- Called debit
>Jun 29 10:59:41 WARNING[8862]: chan_h323.c:588 oh323_write: Asked to
>transmit frame type 4, while native formats is 256 (read/write = 4/4)
>Jun 29 10:59:41 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
>path from g729 to slin
>Jun 29 10:59:41 WARNING[8862]: indications.c:99 playtones_alloc: Unable to
>set 'H323/ip$64.243.115.153:32971/11679' to signed linear format (write)
>voip*CLI>
><-- SIP read from 69.115.205.168:4152:
>SIP/2.0 100 Trying
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>To: <sip:debit at 192.168.15.175:4146>
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 102 INVITE
>User-Agent: Grandstream BT100 1.0.5.16
>Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
>Content-Length: 0
>
>
>--- (9 headers 0 lines)---
>voip*CLI>
><-- SIP read from 69.115.205.168:4152:
>SIP/2.0 180 Ringing
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>To: <sip:debit at 192.168.15.175:4146>;tag=2cfc88182690d7d1
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 102 INVITE
>User-Agent: Grandstream BT100 1.0.5.16
>Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
>Content-Length: 0
>
>
>--- (9 headers 0 lines)---
> -- SIP/debit-9f37 is ringing
>voip*CLI>
><-- SIP read from 69.115.205.168:4152:
>SIP/2.0 200 OK
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK5aab56d3;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>To: <sip:debit at 192.168.15.175:4146>;tag=2cfc88182690d7d1
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 102 INVITE
>User-Agent: Grandstream BT100 1.0.5.16
>Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
>Contact: <sip:debit at 69.115.205.168:4146>
>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>Content-Type: application/sdp
>Supported: replaces
>Content-Length: 213
>
>v=0
>o=debit 0 8000 IN IP4 69.115.205.168
>s=SIP Call
>c=IN IP4 69.115.205.168
>t=0 0
>m=audio 4192 RTP/AVP 4 101
>a=sendrecv
>a=rtpmap:4 G723/8000
>a=ptime:30
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-11
>
>--- (13 headers 11 lines)---
>Found RTP audio format 4
>Found RTP audio format 101
>Peer audio RTP is at port 69.115.205.168:4192
>Found description format G723
>Found description format telephone-event
>Capabilities: us - 0x10d (g723|ulaw|alaw|g729), peer - audio=0x1
>(g723)/video=0x0 (nothing), combined - 0x1 (g723)
>Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
>(telephone-event), combined - 0x1 (telephone-event)
>Jun 29 10:59:46 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
>path from g723 to ulaw
>Jun 29 10:59:46 NOTICE[8862]: channel.c:1893 set_format: Unable to find a
>path from g723 to ulaw
>list_route: hop: <sip:debit at 69.115.205.168:4146>
>set_destination: Parsing <sip:debit at 69.115.205.168:4146> for address/port to
>send to
>set_destination: set destination to 69.115.205.168, port 4146
>Transmitting (NAT) to 69.115.205.168:4152:
>ACK sip:debit at 69.115.205.168:4146 SIP/2.0
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK21677c00;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>o: <sip:debit at 69.115.205.168:4146>;tag=2cfc88182690d7d1
>Contact: <sip:7323600296 at 64.243.115.157>
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 102 ACK
>User-Agent: Asterisk PBX
>Content-Length: 0
>
>
>---
> -- SIP/debit-9f37 answered H323/ip$64.243.115.153:32971/11679
>Jun 29 10:59:46 WARNING[8862]: channel.c:2317 ast_channel_make_compatible:
>No path to translate from H323/ip$64.243.115.153:32971/11679(256) to
>SIP/debit-9f37(1)
>Jun 29 10:59:46 WARNING[8862]: app_dial.c:1324 dial_exec_full: Had to drop
>call because I couldn't make H323/ip$64.243.115.153:32971/11679 compatible
>with SIP/debit-9f37
>set_destination: Parsing <sip:debit at 69.115.205.168:4146> for address/port to
>send to
>set_destination: set destination to 69.115.205.168, port 4146
>Reliably Transmitting (NAT) to 69.115.205.168:4152:
>BYE sip:debit at 69.115.205.168:4146 SIP/2.0
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK079d2b3d;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>To: <sip:debit at 69.115.205.168:4146>;tag=2cfc88182690d7d1
>Contact: <sip:7323600296 at 64.243.115.157>
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 103 BYE
>User-Agent: Asterisk PBX
>Content-Length: 0
>
>
>---
> == Spawn extension (default, 19087773456, 1) exited non-zero on
>'H323/ip$64.243.115.153:32971/11679'
>voip*CLI>
><-- SIP read from 69.115.205.168:4152:
>SIP/2.0 200 OK
>Via: SIP/2.0/UDP 64.243.115.157:5060;branch=z9hG4bK079d2b3d;rport
>From: "7323600296" <sip:7323600296 at 64.243.115.157>;tag=as492d969f
>To: <sip:debit at 192.168.15.175:4146>;tag=2cfc88182690d7d1
>Call-ID: 0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157
>CSeq: 103 BYE
>User-Agent: Grandstream BT100 1.0.5.16
>Warning: 399 69.115.205.168 "detected NAT type is symmetric NAT"
>Contact: <sip:debit at 69.115.205.168:4146>
>Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
>Supported: replaces
>Content-Length: 0
>
>
>--- (12 headers 0 lines)---
>Destroying call '0e4c2a05623aa0dd33497a316edf6671 at 64.243.115.157'
>voip*CLI> sip no debug
>SIP Debugging Disabled
>voip*CLI>
>
>Alex
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stefan
>Gofferje
>Sent: Wednesday, June 29, 2005 10:28 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] Play an announcement to the CALLING party
>
>Hi folks,
>
>how could I play an announcement to the calling party as soon, as the
>called party picked up. I would like to deploy an asterisk in an
>environment, where a premium rate support-number is offered to customers
>which do not want to pay a monthly support contract. In Germany, you are
>commited by law to announce the cost per minute of a premium rate number at
>the beginning of the call. So, to avoid the employees forgetting it, an
>automatic announcement should be played. Besides, same rules are applicable
>for calls that may be recorded for quality assurance issues.
>At least for premium rate calls, queues won't work as the customer would
>strongly dislike hearing an announcement about the rate while waiting for
>an agent.
>The a() option of the dial app only works for CALLED parties and when
>trying to use a macro with the m() option, the Playback also goes to the
>called party.
>Anyone any hints on that?
>
>Regards,
>Stefan
>
>
>
>
--
Ronald Wiplinger (CEO of ELMIT)
http://www.elmit.com +886 (0) 939--77-55-16 or FWD 511208
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