SIP call success stories, anyone? (R57/58_improve, A910)

[motovoip]

Hi all,

finally I got my forum registration issues killed (_very_ nice and fast moderator here!), thus I'd like to ask:

Did anyone succeed in registering this absolutely marvelous modded firmware to a SIP provider?
I succeed in associating to several WLAN networks no prob,
thus the desktop's service registration string for WLAN does appear (plus usable Opera), dito then for any SIP provider I tried, full registration string on desktop, but whichever way I configure it, whenever I try to actually make a SIP call I get "call failed" (possibly accompanied by some screeching sound).
Note that no SIP/RTP network traffic whatsoever can be recorded during this attempt.
I tried both two very different asterisk 1.2 installations and an external SparVoIP account, no go.
Both the phone and asterisks think that the peer is registered, but an actual call attempt fails, despite tweaking all kinds of imaginable and unimaginable protocol settings.

Some hints about getting SIP service showing up as registered: if one fails to register the service, then I usually manage to refresh it successfully by "switching off" WLAN and "quick connect" WLAN again, then it usually works.
If it doesn't, then one has to go as far as rebooting the entire device. Yes, that stuff _is_ buggy. Simply trying to trigger an update via the SIP profile page is NOT reliable.

I then decided to do a USBNET login, and found the following data about the SIP/IMS stack:

/etc/rc.d/rc2.d/S45tapisrv.sh:/usr/SYSqtapp/ims/imsstart.sh &
/usr/SYSqtapp/ims/imsstart.sh starts /etc/ims/proglist
/usr/SYSqtapp/ims/voip_env.sh very interesting!!!
imsstart.sh -C all kills all daemons
/ezxlocal/voip/##sipstackinit.xml
VOIP profile XMLs handled by /usr/data_resource/move/ims/imsvoipuad

I then copied imsstart.sh to the tmp directory, modified it to NOT overwrite the copied proglist, adapted proglist to include -debug parameters (to try to get logfiles), stopped all IMS daemons and started them via imsstart.sh, NOTHING, no logging. Logging would be immensely helpful in this situation, however it doesn't work and I'm afraid that it's because this might be a release build of those binaries and they simply chose to remove all logging functionality in release build configurations, given that this is a limited-space handset after all...

Tried adapting all sorts of SIP / IMS parameters in those XML files, no go, never a successful call after registration.

Just imagine if one managed to do this - then manual-install of some XScale-built vpnc, and there you go with full VoIP functionality wherever you go.

I now have an asterisk 1.4, possibly that one works better once I get to that location to try it.

The users on France Neuf forum are also complaining quite a lot about spotty SIP service, thus the SIP stack itself does seem to be problematic and it's not too astonishing that I'm having sub-zero degrees of success.

The way it's failing for me is that in case of a qualify:d asterisk SIP peer (i.e. asterisk is configured to qualify via OPTIONS requests), I always get "486 Busy Here", and the same 486 for any attempt to dial this phone.
IOW, _something_ is preventing the phone from actively taking part in SIP communication. And no, the phone isn't switched to "Away" setting (possibly resulting in Busy Here) either (GSM works properly, anyway). Networking setup should be fully trouble-free, too, minimized and NAT-related issues locally.

Any other ideas or comments?

Maybe one idea would be to install siproxd to try to connect to a SIP _proxy_, not a _registrar_ (asterisk).

Using R57 V7, BTW.

Thanks!

[sabrod]

I don't know more about sip, but i think it's due to the lack of open codecs of the phone libs.
By greping the neuf's fw, i only found G729, and G723...

[motovoip]

I've been digging through a couple dozen cubic kilometres of Neuf Forum postings (it certainly helps to have had an advanced course of French way back in school....), and I got somewhat further in my understanding, but it didn't help.

Relevant URLs:
Forum Neuf Mobile • Index
Forums Justneuf.com
Forum Neuf Mobile • Voir le sujet - updateProfile pour Linux
Assistance : TWIN
TWIN Motorola A910i
Kit de connexion version PC

Neuf Forum user "newdata27" managed to get his A910i to work using those el-cheapo Betamax provider VoIP domains!

A910i SIP configuration (for Neuf) is said to be done via the 9install Win32 app (see users "DA SILVA" and "sidecar33") which appears to create a file updateProfile.conf. (however the 9install app appears to be available to Neuf subscribers only, would need to ask those people)

User "lufer" described the process as
cd leRepertoireQuiVaBien
cp updateProfile.conf /Volume/MYPHONE/
cd

IOW, the ONLY thing you have to do is create a custom updateProfile.conf with your SIP settings, move it to the mounted microSD card (which should have been formatted by the phone itself!! they say that makes a difference; plus there are provider notes that SIP _needs_ the SD card to work properly!), and the phone should update its VoIP entries.
But, well... for me there are no changes at all so far.

Someone said that the Wifi sections in the updateProfile.conf prevented the SIP update from working, and once removing the Wifi sections it worked. That fact might point at some kind of syntax problem in my SIP-section-only updateProfile.conf .

Also, I'm not sure which CRLF mode updateProfile.conf should have, or whether that's even relevant.

At first I doubted that updateProfile.conf is relevant for the A910i phones (there are other phones which need that), but then I verified that several users listed that file in combination with A910i, thus that seems to be needed.

I'll investigate a little bit more, but definitely not much since I'm semi-fed-up now.

[sabrod]

Hey dude! fully admirating your research! I just want to propose my translation help, as a french...

I just wanted to say, if i was in the hurry, i would try to produce an alternative (linphone, asterix based client..., even j2me sip client...) because you could find a solution, but you could perfectly face a VERY weird stuff and lost a lot of hours boring to try to configure something...

Also i could download 9install. (i can't remember, where... ) So i know it's just supposed to copy a configuration, and also contain some "automatic secure setting" i mean, only related to the way neuf send your id/password from their website, to your computer running 9install, then the user don't have to care about his configuration, he just launch the 9install while connected to internet...

Neuf is a DSL provider ,mainly, but he used the A910i as it's first SIP / GSM "twin" phone, to provide (as a virtual operator) diverse other Voip/Gsm services.
Maybe the forum moved since i went, but i remember these crappies admins that was lying, etc... (you know a provider is never of any help when a user come to ask technical stuffs).
Then, they used also E28 phones... They never unlocked any a910i afaik... (they sold several locked a910i's but always tell craps on phone , to refuse the unlock code ??
A910i is a cool phone, it's maybe only the SIP codecs that are in uses, but i never really tried this intensively, i gave up as well I think i will build some sip alternative software instead. :/
Qtopia 4 OS could run very well on A910, too...So why bother that much with the motorola dark technology?

[motovoip]

I just wanted to say, if i was in the hurry, i would try to produce an alternative (linphone, asterix based client..., even j2me sip client...) because you could find a solution, but you could perfectly face a VERY weird stuff and lost a lot of hours boring to try to configure something...

Indeed, the problem with this setup is that it's entirely undocumented/unacessible and you don't have any idea where to start debugging this.

A910i is a cool phone, it's maybe only the SIP codecs that are in uses, but i never really tried this intensively, i gave up as well I think i will build some sip alternative software instead. :/
Qtopia 4 OS could run very well on A910, too...So why bother that much with the motorola dark technology?

After this experience I'm now much more willing to reinvest time into openezx things. After all they (well, "we") got lots of things working now, and environment integration is just around the corner, and possibly one could add a nice, full-featured SIP stack then and get it to work with remotely comparable effort even.

Problem is that environment fine-tuning is what is really difficult I think, getting an initial environment semi-working should work, but semi-perfection would take a long time.

[sabrod]

Also for stracing or lsof, we shoul dneed a "working" scheme... someone with neuf operator, and an A910 with that SIP setup...

Heh i totally agree.
You know, one of the first touchscreen device, the Apple newton, was though 10 years before its first sell! (and it is amazing, but was released too early, never really got success because it was too modern for its time )

I believe we "should" have a multiboot system, with Qtopia and SHR/FSO, one to use as far as it can be used... The other to work on...
Qtopia is a good system, but i find it too simple, too "austere" , not enough services integration, linkage between relaed apps...

But qtopia can help us to develop apps in C, for a GNU rootfs, use our phone with wifi and wifi, use it as a cordless phone at home, etc... Well, whatever we want

The only prob on A910, is the mmc_spi.c driver that is recurently broken... But it's just lack of time as a lot of people use it, there is probably patches around the net...
(i uploaded an A910 "reference" (means working) kernel, in my people account at openezx). Not that it is hard to make one, but you can compare, at least, you also have the git hash to build from the same tree. Unfortunatly, i don't think the BP handshake is properly done.
But that shouldn't differ a lot from the other phones... It's probably the USB driver.

[sabrod]

Ah... if you need to run and have VOIP, maybe you can beat OpenEZx at it, using our 2.4 base OABI stock OS...

I found this on opentwin.org (or their ML...)

(SIP configs files for another neuf's phone, and it's example for implementing another SIP account too)...
Here it is:

Voila
des cat des fichiers de configs qui vont bien pour les SIP free et 9
sur le E2831 smarCore :)
/usr/share/config/sip # cat neuf
[audio_codec_0]
mime=PCMA
[audio_codec_1]
mime=PCMU
[audio_codec_2]
mime=G729
[net]
use_nat=0
[rtp]
audio_rtp_port=8080
ptime=20
audio_jitt_comp=500
[stun]
stun_server=64.69.76.23
use_stun=0
[echo_cancel]
enabled=0
[agc]
enabled=0
[ext_auth_program]
enabled=0
[dtmf]
send_method=21
[CONFERENCE]
conference_payload=0
[AUTH]
remove_auth_for_register=0
[DNS_LOOKUP]
after_reg_timeout=1
[MWI]
VM4USER=123
VM4SERVER=123
[REFER]
enabled=0
unconditional=0
on_busy=0
no_answer=0
refer_to=sip:
[sip]
sip_port=5060
contact=sip:LE_TELEPHONE@neuf.com
domain=neuf.com
use_info=1
guess_hostname=0
pre_cp_tone_sending=1
sip_ping_sending=1
call_no_answer_timeout=30
sip_ping_refresh_timer=55
need_to_chk_wifi_link_up=1
retransmit_200ok=0
retransmit_ack=0
send_de_register=0
preferred_codec_only=1
[proxy_0]
reg_proxy=sip:proxy.sip.n9uf.net:5060
reg_route=
reg_identity=sip:LE_TELEPHONE@neuf.com
reg_expires=4800
reg_sendregister=1
registrar=sip:neuf.com
[auth_info_0]
username=LE_TELEPHONE
userid=LE_TELEPHONE
passwd=LE_MOT_DE_PASSE
ha1=md5
realm=myrealm.com
[EXPIRES]
changable=1
[DEFAULT_TIME]
DEFAULT_T1=500
DEFAULT_T2=4000
DEFAULT_T4=5000
REG_RETRY_TIME_L=90
[HANDOVER]
config_manual=0
gsm_handover=
voip_handover=
signal_drop=
gsm_psi=
sms_psi=
[FILENAME]
profilename=neuf
et pour Celui de free:
/usr/share/config/sip # cat free
[audio_codec_0]
mime=PCMU
[audio_codec_1]
mime=PCMA
[audio_codec_2]
mime=G729
[net]
use_nat=0
[rtp]
audio_rtp_port=8080
ptime=20
audio_jitt_comp=500
[stun]
stun_server=64.69.76.23
use_stun=0
[echo_cancel]
enabled=0
[agc]
enabled=0
[ext_auth_program]
enabled=0
[dtmf]
send_method=2
[CONFERENCE]
conference_payload=0
[AUTH]
remove_auth_for_register=0
[DNS_LOOKUP]
after_reg_timeout=1
[MWI]
VM4USER=123
VM4SERVER=123
[REFER]
enabled=0
unconditional=0
on_busy=0
no_answer=0
refer_to=sip:
[sip]
sip_port=5060
contact=sip:LE_TELEPHONE@freephonie.net
domain=freephonie.net
use_info=1
guess_hostname=0
retransmit_200ok=0
retransmit_ack=0
send_de_register=0
preferred_codec_only=1
[proxy_0]
reg_proxy=sip:freephonie.net:5060
reg_route=
reg_identity=sip:LE_TELEPHONE@freephonie.net
reg_expires=3600
reg_sendregister=1
registrar=sip:freephonie.net
[auth_info_0]
username=LE_TELEPHONE
userid=LE_TELEPHONE
passwd=LE_MOT_DE_PASSE
ha1=md5
realm=myrealm.com
[EXPIRES]
changable=1
[DEFAULT_TIME]
DEFAULT_T1=500
DEFAULT_T2=4000
DEFAULT_T4=5000
REG_RETRY_TIME_L=90000
[HANDOVER]
config_manual=0
gsm_handover=
voip_handover=
signal_drop=
gsm_psi=
sms_psi=
[FILENAME]
profilename=free
Il reste a changer les champs: LE_TELEPHONE & LE_MOT_DE_PASSE par
les données correspondantes...voilà ce n'est peut etre pas des réglages
parfaits ! mais ca a le mérite de marcher très bien :D

But regarding the great improvements of openezx, i don't want to work in the "wrong way" ...

Have fun!