RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?
I did some packet sniffs, and below are two sets of packets, the first is the second phone line that works fine with an incoming call and outgoing sound This seems to be the key packet that sets up the codes and sessions ( I really don't know any of this sip stuff well, but hopefully somebody on the list knows it): The main thing to point out is the initial "Media Description" section. In the Working line2, it's: Media Description, name and address (m): audio 16446 RTP/AVP 0 101 ... Media Format: 101 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-15 So I believe this is setting up the audio transmit stuff. On the line1, this data is NOT being sent. I don't know what this stuff means, still looking into it..but maybe someone here does know it? Am I possibly even on the right track?? The other thought I have, since this is data being sent FROM the Sipura TO asterisk, the problem is once again seeming to point directly at Sipura, and it's basically not sending the audio info.. Does any of this even make any sense?? Hope this either helps others to possibly find a fix, or if anyone _does_ have a fix, please let me know! Packet for line2, working outgoing audio Frame 7 (733 bytes on wire, 733 bytes captured) Ethernet II, Src: 00:0e:08:aa:b7:b1, Dst: 00:07:95:55:7b:ce Internet Protocol, Src Addr: 192.168.1.21 (192.168.1.21), Dst Addr: 192.168.1.20 (192.168.1.20) User Datagram Protocol, Src Port: 5061 (5061), Dst Port: 5060 (5060) Session Initiation Protocol Status line: SIP/2.0 200 OK Status-Code: 200 Message Header To: ;tag=6b4e39bb53bc50bc SIP to address: SIP tag: 6b4e39bb53bc50bc From: "asterisk" ;tag=as55a02558 SIP from address: "asterisk" SIP tag: as55a02558 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK0d8c5d0f Contact: SPA 2202 Server: Sipura/SPA2000-2.0.2 Content-Length: 210 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 25669620 25669620 IN IP4 192.168.1.21 Owner Username: - Session ID: 25669620 Session Version: 25669620 Owner Network Type: IN Owner Address Type: IP4 Owner Address: 192.168.1.21 Session Name (s): - Connection Information (c): IN IP4 192.168.1.21 Connection Network Type: IN Connection Address Type: IP4 Connection Address: 192.168.1.21 Time Description, active time (t): 0 0 Session Start Time: 0 Session Start Time: 0 Media Description, name and address (m): audio 16446 RTP/AVP 0 101 Media Type: audio Media Port: 16446 Media Proto: RTP/AVP Media Format: 0 Media Format: 101 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute Fieldname: rtpmap Media Attribute Value: 101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute Fieldname: fmtp Media Attribute Value: 101 0-15 Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): sendrecv Below is an incoming phone call to line1, with the outgoing voice NOT working: Frame 7 (672 bytes on wire, 672 bytes captured) Ethernet II, Src: 00:0e:08:aa:b7:b1, Dst: 00:07:95:55:7b:ce Internet Protocol, Src Addr: 192.168.1.21 (192.168.1.21), Dst Addr: 192.168.1.20 (192.168.1.20) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status line: SIP/2.0 200 OK Status-Code: 200 Message Header To: ;tag=f03d01bbf25c28bb SIP to address: SIP tag: f03d01bbf25c28bb From: "asterisk" ;tag=as5c261e75 SIP from address: "asterisk" SIP tag: as5c261e75 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK7b7c931a Contact: Ext 2201 Server: Sipura/SPA2000-2.0.2 Content-Length: 154 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIO
RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?
I don't know if this helps, but I started having this problem after I sent out a fax. My fax machine was connected to line 1 at that time. I tried changing the FAX detection settings but no luck. Regards, Victor Perez -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Musone Sent: Sunday, April 18, 2004 10:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem? I'm not sure if anybody has determined the cause/fix for this problem, but I am getting the same problem. I turned on syslog debugging and there were some interesting results: ... My feeling is that this is a Sipura problem. I've upgraded to the firmare 2.02, but still no difference. ___ 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] Sipura line 1 outgoing voice problem?
host=dynamic context=home secret=XXXx callerid="SPA2" <2202> mailbox=2202 dtmfmode=rfc2833 canreinvite=no nat=0 [2201] type=friend host=dynamic context=home secret=Xxxx callerid="SPA1" <2201> mailbox=2202 dtmfmode=rfc2833 canreinvite=no nat=0 I'm also sending a copy of this to the sipura tech support.. Hope this either helps others to possibly find a fix, or if anyone _does_ have a fix, please let me know! I'm pretty close to just returning it, as to me, this simply does not work. Thanks! -Mark - Original Message - From: Matt McIntyre To: [EMAIL PROTECTED] Sent: Tuesday, March 23, 2004 6:59 PM Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem? I am experiencing this same problem and was wondering if anyone has come to a resolution. I have contacted Sipura but have not heard any response yet and am having trouble determining for sure whether the problem resides with Asterisk or the Sipura. As I have noticed that there are many users on the list who use the Sipura unit without this problem (and even a fellow with one unit that worked and one that did) I think the Sipura must be suspect. __ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) ___ 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] Sipura line 1 outgoing voice problem?
Maybe this helps. I have 4 sipuras on the same network as Asterisk. I had to make sure each line on the sipura uses a different sip port: 5060/5061 on the first one, 5062/5063 on the second, and so on. Best regards, - Original Message - From: Matt McIntyre To: [EMAIL PROTECTED] Sent: Tuesday, March 23, 2004 6:59 PM Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem? I am experiencing this same problem and was wondering if anyone has come to a resolution. I have contacted Sipura but have not heard any response yet and am having trouble determining for sure whether the problem resides with Asterisk or the Sipura. As I have noticed that there are many users on the list who use the Sipura unit without this problem (and even a fellow with one unit that worked and one that did) I think the Sipura must be suspect. __ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) ___ 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] Sipura line 1 outgoing voice problem?
I am experiencing this same problem and was wondering if anyone has come to a resolution. I have contacted Sipura but have not heard any response yet and am having trouble determining for sure whether the problem resides with Asterisk or the Sipura. As I have noticed that there are many users on the list who use the Sipura unit without this problem (and even a fellow with one unit that worked and one that did) I think the Sipura must be suspect. Thanks, Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, March 17, 2004 11:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sipura line 1 outgoing voice problem? __ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) NAT is not an issue as the Sipura and * are on the same network. Is anyone else having this problem? It looks like other people are using Sipura (I saw one user with 30 of them ?!) and am surprised that nobody else is complaining about this problem. I am willing to step through some sip debug if anyone is interested in the output. * version: Asterisk CVS-02/08/04-22:22:57 Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem would go away) Relevent config sections: --8<-- sip.conf --8<-- [cordless1] type=friend username=cordless1 secret=xxx host=dynamic context=cordless1 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw [cordless2] type=friend username=cordless2 secret=xxx host=dynamic context=cordless2 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw -- Chris ___ I had the exact same problem with a Mediatrix 1102doing a flash hook brought both sides of the conversation together. I found out that my sip.conf file had GSM as the first priority codec and the 1102 doesn't support GSM. I kept that the same but put a "disallow = gsm" statement in my sip entry for the 1102 so g.711ulaw would be the first negotiated codec. That fixed the problem. VZ
Re: [Asterisk-Users] Sipura line 1 outgoing voice problem?
__ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) NAT is not an issue as the Sipura and * are on the same network. Is anyone else having this problem? It looks like other people are using Sipura (I saw one user with 30 of them ?!) and am surprised that nobody else is complaining about this problem. I am willing to step through some sip debug if anyone is interested in the output. * version: Asterisk CVS-02/08/04-22:22:57 Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem would go away) Relevent config sections: --8<-- sip.conf --8<-- [cordless1] type=friend username=cordless1 secret=xxx host=dynamic context=cordless1 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw [cordless2] type=friend username=cordless2 secret=xxx host=dynamic context=cordless2 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw -- Chris ___ I had the exact same problem with a Mediatrix 1102doing a flash hook brought both sides of the conversation together. I found out that my sip.conf file had GSM as the first priority codec and the 1102 doesn't support GSM. I kept that the same but put a "disallow = gsm" statement in my sip entry for the 1102 so g.711ulaw would be the first negotiated codec. That fixed the problem. VZ
RE: [Asterisk-Users] Sipura line 1 outgoing voice problem?
Chris Higgins wrote: > Back in January I started having a problem with my Sipura (and there > was > at least one other on the list with the same problem) that if I answer > an incoming call (via X100P) on line 1 of my Sipura, the caller cannot > hear any voice from the internal extension. If the internal user puts > the external user on hold (via flash hook) and returns, both > directions > of audio are fine. I have not had this problem... And I use X100P as well in same setup. BUT... There are other problems I have or had. > > [cordless1] > type=friend > username=cordless1 > secret=xxx > host=dynamic > context=cordless1 > dtmfmode=info > mailbox=1234 > canreinvite=no > disallow=all > allow=alaw > If you are using your SPA 2000 directly with * maybe it is better to have "canreinvite" set to yes. ??? Also.. I think that auto default dtmfmode for SPA is AVT (which is RFC2833)... So check that.!!! ___ 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] Sipura line 1 outgoing voice problem?
Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) NAT is not an issue as the Sipura and * are on the same network. Is anyone else having this problem? It looks like other people are using Sipura (I saw one user with 30 of them ?!) and am surprised that nobody else is complaining about this problem. I am willing to step through some sip debug if anyone is interested in the output. * version: Asterisk CVS-02/08/04-22:22:57 Sipura firmware: 1.0.31 (just upgraded tonight to see if the problem would go away) Relevent config sections: --8<-- sip.conf --8<-- [cordless1] type=friend username=cordless1 secret=xxx host=dynamic context=cordless1 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw [cordless2] type=friend username=cordless2 secret=xxx host=dynamic context=cordless2 dtmfmode=info mailbox=1234 canreinvite=no disallow=all allow=alaw -- Chris ___ 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
